0.10.24.2 pre-release

[platform/upstream/gstreamer.git] / ChangeLog

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

=== release 0.10.24 ===

2009-08-05 00:56:58 +0100  Jan Schmidt <thaytan@noraisin.net>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/gst-plugins-base-plugins.hierarchy:
        * docs/plugins/gst-plugins-base-plugins.interfaces:
        * docs/plugins/gst-plugins-base-plugins.prerequisites:
        * docs/plugins/gst-plugins-base-plugins.signals:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-app.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gdp.xml:
        * docs/plugins/inspect/plugin-gio.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * 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

2009-08-05 00:38:40 +0100  Jan Schmidt <thaytan@noraisin.net>

        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/lv.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/tr.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          Update .po files

2009-08-01 17:26:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/typefind/gsttypefindfunctions.c:
        * tests/check/gst/typefindfunctions.c:
          typefinding: fix detection of fLaC id packet in broken flac-in-ogg
          There are flac-in-ogg files without the usual flac packet framing
          and these files just have a 4-byte fLaC ID packet as first packet.
          We need to recognise the type just from these four bytes if we
          want oggdemux to recognise these streams correctly.

2009-07-30 14:40:50 +0100  Jan Schmidt <thaytan@noraisin.net>

        * ChangeLog:
        * configure.ac:
        * 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

2009-07-29 14:15:53 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

        * 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.

2009-07-30 13:42:21 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.

2009-07-30 09:28:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-07-27 10:29:27 +0100  Balachandran C <balachandran_c@rediffmail.com>

        * ext/alsa/gstalsasrc.c:
          alsasrc: set alsasrc->handle back to NULL when closing device
          Fixes crashes in gst_alsa_find_device_name() when probing or
          reading the device-name property (e.g. when doing a dot-file
          dump). Fixes #589797.

2009-07-24 19:26:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-07-24 13:39:55 +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/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

2009-07-24 13:46:15 +0100  Jan Schmidt <thaytan@noraisin.net>

        * tests/examples/v4l/.gitignore:
          ignores: Ignore v4l probing example binary

2009-07-24 09:35:38 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-07-24 00:42:16 +0300  Stefan Kost <ensonic@users.sf.net>

        * common:
          Automatic update of common submodule
          From fedaaee to 94f95e3

2009-07-22 14:21:43 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

        * gst-plugins-base.spec.in:
          Update spec file with latest changes

2009-07-20 17:28:20 +0100  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/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

2009-07-20 12:51:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-07-03 04:42:24 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

        * gst-libs/gst/audio/gstaudiosink.c:
          audiosink: Add stream-status messages
          Fixes #587695

2009-07-03 04:41:05 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

        * gst-libs/gst/audio/gstaudiosrc.c:
          audiosrc: Add stream-status messages
          See #587695

2009-07-20 10:53:11 +0200  Edward Hervey <bilboed@bilboed.com>

        * gst/adder/gstadder.c:
          gstadder: Don't forget to free pending events on flush/dispose.
          Fixes #588747

2009-07-12 10:08:12 +0200  Edward Hervey <bilboed@bilboed.com>

        * tests/check/elements/adder.c:
          tests/adder: Add stream consistency checking. Fixes #588748

2009-07-12 10:07:34 +0200  Edward Hervey <bilboed@bilboed.com>

        * 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.

2009-07-13 09:28:54 +0200  Edward Hervey <bilboed@bilboed.com>

        * gst/adder/gstadder.c:
        * gst/adder/gstadder.h:
          adder: Collect incoming tag events and send them after newsegment. Fixes #588747

2009-07-16 09:32:46 +0200  Edward Hervey <bilboed@bilboed.com>

        * ext/vorbis/vorbisdec.c:
          vorbisdec: Check for empty tag strings. Fixes #588724

2009-07-14 17:03:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-07-15 17:40:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-07-15 17:26:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-07-14 19:00:36 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

        * tests/examples/v4l/Makefile.am:
          examples: fix v4l probe example build
          Fixes bug #588550.

2009-07-14 19:00:10 +0100  Jan Schmidt <thaytan@noraisin.net>

        * ChangeLog:
        * configure.ac:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/tr.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          0.10.23.2 pre-release

2009-07-14 16:24:10 +0100  Jan Schmidt <thaytan@noraisin.net>

        * po/LINGUAS:
        * po/tr.po:
          Add Turkish translations

2009-07-14 15:31:13 +0100  Jan Schmidt <thaytan@noraisin.net>

        * tests/check/elements/adder.c:
          adder: One more attempt to fix the adder test
          Give up and discard and recreate the alsasrc after checking it can
          be opened, due to some strange crash inside alsa when we don't.

2009-07-14 15:06:41 +0100  Jan Schmidt <thaytan@noraisin.net>

        * tests/check/elements/adder.c:
          adder: Perform get_state() in the unit test
          Wait for the alsasrc to return to NULL after setting it to PAUSED for
          testing, otherwise it leads to segfaults later on.

2009-07-14 14:39:32 +0100  Jan Schmidt <thaytan@noraisin.net>

        * 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.

2009-07-13 22:51:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-02-21 13:18:10 +0000  Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>

        * ext/ogg/gstoggmux.c:
          oggmux: add Kate caps to the list of accepted types
          See #525743.

2009-07-13 21:56:46 +0300  Stefan Kost <ensonic@users.sf.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://...

2009-07-13 21:54:47 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/interfaces/navigation.c:
          navigation: simplify docs
          Make short-desc short - its used in the toc. Strip uneeded markup.

2009-07-13 18:31:15 +0100  Jan Schmidt <thaytan@noraisin.net>

        * win32/common/libgstnetbuffer.def:
        * win32/common/libgstvideo.def:
          win32: Fix exports
          Remove methods from video base classes that have moved to -bad.
          Add gst_netaddress_to_string

2009-07-13 17:56:58 +0100  Jan Schmidt <thaytan@noraisin.net>

        * tests/examples/gio/.gitignore:
          ignores: ignore the giosrc-mounting example binary

2009-07-13 17:54:40 +0100  Jan Schmidt <thaytan@noraisin.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.

2009-07-13 17:52:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/pbutils/descriptions.c:
          pbutils: add description for Siren codec and make two descriptions non-translatable

2009-07-13 12:23:20 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From 5845b63 to fedaaee

2009-07-13 18:21:49 +0200  Elliott Sales de Andrade <quantum.analyst at gmail.com>

        * 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.

2009-07-13 14:55:59 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

        * 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

2009-07-03 11:38:01 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>

        * configure.ac:
        * sys/v4l/Makefile.am:
        * sys/v4l/gstv4lelement.c:
          v4lsrc: optional support for device probing with gudev
          Enumerate v4l devices using gudev if available.
          Fixes bug #583640.

2009-07-10 23:24:36 +0100  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: add since tags to docs

2009-07-10 21:29:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-07-10 21:01:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-07-10 20:25:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/seek.c:
          seek: add a checkbox for progressive download

2009-07-10 20:24:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-07-10 20:04:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-07-10 19:59:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-07-10 19:49:46 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-07-10 20:06:28 +0100  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-07-10 18:59:05 +0100  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-07-10 19:16:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiosink.c:
          gio: Also post a "not-mounted" message from giosink

2009-07-10 17:15:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * tests/examples/gio/giosrc-mounting.c:
          gio: Remove workaround for playbin2 bug in the sample application
          The playbin2 bug was #588078.

2009-07-10 17:08:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

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

2009-07-10 11:24:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * configure.ac:
          gio: Remove the experimental status from the GIO plugin
          Fixes bug #510417.

2009-07-10 11:24:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiosink.c:
        * ext/gio/gstgiosrc.c:
          gio: Add documentation for the new "not-mounted" and "file-exists" messages

2009-07-09 13:45:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiobasesrc.c:
          gio: Make sure that we have the correct stream position when starting

2009-07-08 17:24:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

2009-07-08 17:19:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiobasesink.c:
        * ext/gio/gstgiobasesink.h:
        * ext/gio/gstgiobasesrc.c:
        * ext/gio/gstgiobasesrc.h:
        * ext/gio/gstgiosink.c:
        * ext/gio/gstgiosrc.c:
          gio: Don't close the GIO streams for the giostream{src,sink} elements
          This makes it possible to do something useful with the streams
          after the element has stopped. Fixes bug #587896.

2009-07-08 17:19:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * tests/check/pipelines/gio.c:
          gio: Try to reuse the pipeline with the same stream objects

2009-07-08 17:02:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiobasesink.c:
        * ext/gio/gstgiobasesrc.c:
          gio: Improve the error message if a stream is already closed before usage

2009-07-08 16:55:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

2009-07-08 16:54:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiosrc.c:
          gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted

2009-07-08 16:50:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiosink.c:
          gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink

2009-07-08 15:52:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

2009-07-08 15:08:32 +0200  Philip Jägenstedt <philipj@opera.com>

        * gst/audioconvert/gstchannelmix.c:
          audioconvert: Fix compilation when debugging is disabled
          Fixes bug #587980.

2009-07-07 20:23:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiobasesink.c:
        * ext/gio/gstgiobasesink.h:
        * ext/gio/gstgiobasesrc.h:
        * ext/gio/gstgiosink.c:
        * ext/gio/gstgiosink.h:
        * ext/gio/gstgiostreamsink.c:
        * ext/gio/gstgiostreamsink.h:
          gio: Add vfunc for requesting the stream for the sinks too

2009-07-07 20:21:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiobasesink.c:
        * ext/gio/gstgiobasesink.h:
        * ext/gio/gstgiobasesrc.c:
        * ext/gio/gstgiosink.c:
        * ext/gio/gstgiosrc.c:
        * ext/gio/gstgiostreamsink.c:
        * ext/gio/gstgiostreamsrc.c:
          gio: Some more random cleanup

2009-07-07 20:20:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgio.c:
        * ext/gio/gstgiobasesink.c:
        * ext/gio/gstgiobasesrc.c:
        * ext/gio/gstgiobasesrc.h:
        * ext/gio/gstgiosink.c:
        * ext/gio/gstgiosrc.c:
        * ext/gio/gstgiosrc.h:
        * ext/gio/gstgiostreamsink.c:
        * ext/gio/gstgiostreamsrc.c:
        * ext/gio/gstgiostreamsrc.h:
          gio: Update my mail address and copyright

2009-07-07 20:18:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/gio/gstgiobasesrc.c:
        * ext/gio/gstgiobasesrc.h:
        * ext/gio/gstgiosrc.c:
        * ext/gio/gstgiostreamsrc.c:
        * ext/gio/gstgiostreamsrc.h:
          gio: General clean up and simplification
          The GInputStreams are now requested by a vfunc from
          the subclasses instead of relying that the subclass
          sets it until it's needed.
          This might also fix bug #587896.

2009-07-06 22:31:12 +0100  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: keep sending newsegments after seeking
          Adder sends with timestamps from 0 upwards. After seeking we need to send
          new-segments to get correct positions-queries.

2009-07-06 20:44:00 +0100  Stefan Kost <ensonic@users.sf.net>

        * tests/check/elements/adder.c:
          adder: make test more robust
          Add audioconverts to the live-seeking test to make it negotiate.

2009-06-30 17:19:50 +0300  Stefan Kost <ensonic@users.sf.net>

        * sys/xvimage/xvimagesink.c:
          xvimagesink: use core performance log category

2009-07-05 21:29:40 +0200  Edward Hervey <bilboed@bilboed.com>

        * gst/adder/gstadder.c:
          adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
          This ensures that collectpads' cookie is properly updated so that when the streaming
          threads will restart and be checking for the flushing status of all pads there will
          be no inconsistent state.

2009-07-05 18:01:38 +0200  Hans-Peter Nilsson <hp@gcc.gnu.org>

        * ext/pango/gstclockoverlay.c:
          pango: Call tzset() before localtime_r()
          POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
          required to set the state variables that define the current timezone.  Indeed,
          glibc (at least 2.9) doesn't do this for subsequent calls.  The effect is that
          if the system timezone is changed for a running program between two calls to
          gst_clock_overlay_render_time, it won't be noticed.  For glibc, changing the
          timezone equals /etc/localtime being modified.
          Fixes bug #587676.

2009-07-01 17:33:14 -0700  David Schleef <ds@schleef.org>

        * ext/Makefile.am:
          build: remove spurious schroedinger reference

2009-07-01 10:25:43 -0700  David Schleef <ds@schleef.org>

        * configure.ac:
        * ext/Makefile.am:
        * ext/schroedinger/Makefile.am:
        * ext/schroedinger/gstschro.c:
        * ext/schroedinger/gstschrodec.c:
        * ext/schroedinger/gstschroenc.c:
        * ext/schroedinger/gstschroparse.c:
        * ext/schroedinger/gstschroutils.c:
        * ext/schroedinger/gstschroutils.h:
        * gst-libs/gst/video/Makefile.am:
        * gst-libs/gst/video/gstbasevideocodec.c:
        * gst-libs/gst/video/gstbasevideocodec.h:
        * gst-libs/gst/video/gstbasevideodecoder.c:
        * gst-libs/gst/video/gstbasevideodecoder.h:
        * gst-libs/gst/video/gstbasevideoencoder.c:
        * gst-libs/gst/video/gstbasevideoencoder.h:
        * gst-libs/gst/video/gstbasevideoparse.c:
        * gst-libs/gst/video/gstbasevideoparse.h:
        * gst-libs/gst/video/gstbasevideoutils.c:
        * gst-libs/gst/video/gstbasevideoutils.h:
          basevideo: send basevideo back to remedial school
          Move basevideo classes and schroedinger plugin to -bad.

2009-07-01 12:54:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-sections.txt:
        * gst-libs/gst/netbuffer/gstnetbuffer.h:
          netaddress: add constant for max len

2009-07-01 12:48:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-sections.txt:
        * gst-libs/gst/netbuffer/gstnetbuffer.c:
        * gst-libs/gst/netbuffer/gstnetbuffer.h:
          netbuffer: add gst_netaddress_to_string
          Add function to serialize a net address to a string.
          API: GstNetAddress::gst_netaddress_to_string()

2009-06-30 18:44:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          uridecodebin: make fd:// uri use buffering too
          fd:// usually operate in push mode only and are thus suitable for buffering.

2009-06-30 14:46:38 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstplaybin2.c:
        * gst/volume/gstvolume.c:
          volume: include "1.0=100%" in property description

2009-06-30 14:45:51 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstplaysink.c:
          playsink: remove unused property defs

2009-06-29 17:11:50 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/audio/multichannel.c:
          multichannel: rewrite the new doc comment a bit
          Its part of the audio lib.

2009-06-29 14:34:02 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst/playback/gstplaysink.c:
          playsink: Avoid a segfault when the video sink fails to start
          Don't attempt to display the subpictures and segfault when the
          video sink failed to start (and hence the videochain is NULL).

2009-06-29 15:14:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstringbuffer.c:
        * gst-libs/gst/audio/gstringbuffer.h:
          ringbuffer: add vmethod to clear the ringbuffer
          Add a vmethod so that subclasses can be notified when they should clear the data
          in the ringbuffer.

2009-06-29 14:00:14 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst-libs/gst/riff/riff-media.c:
          riff-media: Fix the fourcc caps property for VC-1/WMVA
          The caps property for carrying fourccs is 'format', not 'fourcc'

2009-06-29 12:20:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: include in.h for FreeBSD compat
          Fixes #586920

2009-06-29 12:20:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * win32/common/libgstapp.def:
          defs: add defs for new appsink buffer-list method

2009-06-29 12:14:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/app/gstappsink.c:
        * gst-libs/gst/app/gstappsink.h:
          appsink: add docs and signals
          Add docs for the new callback.
          Add signals for the new buffer-list support.

2009-06-29 10:24:36 +0200  Branko Subasic <branko@lnxbranko2.se.axis.com>

        * tests/check/elements/appsink.c:
          Added unit tests for buffer list support in appsink.

2009-06-17 11:12:08 +0200  Branko Subasic <branko@lnxbranko2.se.axis.com>

        * gst-libs/gst/app/gstappsink.c:
          Added buffer list support.

2009-06-17 09:23:11 +0200  Branko Subasic <branko@lnxbranko2.se.axis.com>

        * gst-libs/gst/app/gstappsink.h:
          Added buffer list support.

2009-06-29 09:36:27 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/sdp/gstsdpmessage.c:
          sdp: Include winsock2.h after defining WINVER.
          Similar to bug #587080.

2009-06-29 09:31:40 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Moved a comment.

2009-06-27 23:23:02 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/audio/audio.c:
        * gst-libs/gst/audio/multichannel.c:
          docs: add basic section docs for multichannel and relocate the ones for audio
          Add section docs for multichannel, so that it has a short desc in the toc too.
          Move the section docs in adio up, so that the follow the copyright like
          elsewhere.

2009-06-26 21:11:45 +0300  Stefan Kost <ensonic@users.sf.net>

        * sys/v4l/gstv4lelement.c:
        * sys/v4l/gstv4lsrc.c:
          v4l: open/close device in ready.
          Simillar change like in v4l2src. This allows probing feature in paused, where
          streaming is noit yet started.

2009-06-10 17:05:22 +0300  René Stadler <rene.stadler@nokia.com>

        * gst/playback/gstplaysink.c:
          playbin2: fix initial volume handling also when reusing the element
          This is a follow-up to commit 452988, making it work correctly when the audio
          chain is reused.

2009-06-26 21:48:58 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          Define WINVER before including any win headers
          Fixes bug #587080.

2009-06-27 00:50:54 +0300  René Stadler <mail@renestadler.de>

        * gst-libs/gst/riff/riff-read.c:
          riff: prevent crash if rounded up tag size exceeds data size
          When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
          and an invalid read past the buffer data follows.

2009-06-26 15:17:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/video/gstbasevideocodec.c:
          basevideocodec: By default don't allow caps changes on the srcpad
          This fixed playback of Dirac files with schrodec when upstream wants
          a different width/height, basevideocodec accepts this and then
          pushes buffers with new caps but content of the old caps.
          In the best case this will just result in wrong unit size and a
          failure in basestransform elements.

2009-06-26 14:11:21 +0100  Jan Schmidt <thaytan@noraisin.net>

        * autogen.sh:
          autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
          Check for more automake command variants. Use printf instead of 'echo -n'
          for portability

2009-06-26 13:41:38 +0100  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Automatic update of common submodule
          From f810030 to 5845b63

2009-06-26 13:14:02 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstscreenshot.c:
          screenshot: don't leak message

2009-06-25 12:04:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/typefind/gsttypefindfunctions.c:
          typefinding: lower the h264 typefinder's probability
          A NEARLY_CERTAIN is absolutely not warranted given the kind
          of things it checks for. Even a LIKELY is probably not entirely
          appropriate.

2009-06-24 15:13:56 +0100  Jan Schmidt <jan.schmidt@sun.com>

        * common:
          Automatic update of common submodule
          From f3bb51b to f810030

2009-06-24 09:48:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/pbutils/descriptions.c:
          pbutils: add description for multipart
          So we get slightly nicer error messages when multipartdemux is missing.

2009-06-23 18:07:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/adder/gstadder.c:
          adder: only unflush when we flushed before
          Ass suggested by Stefan Kost:
          Keep track of when the sinkpad was set to flushing and unflush the pad when an
          upstream flushing seek failed.

2009-06-23 15:10:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          uridecodebin: fix leak when the source fails to change state

2009-06-23 12:40:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/subparse/gstssaparse.c:
          ssaparse: avoid leaking all buffers

2009-06-22 22:18:03 +0300  Stefan Kost <ensonic@users.sf.net>

        * tests/check/elements/adder.c:
          adder: test seek handling in adder
          This tests seeking on an adder that has a normal and a live source connected.
          Wheter the current behavior is the desired one needs to be discussed still
          (see #586033)

2009-06-22 16:17:10 +0300  Stefan Kost <ensonic@users.sf.net>

        * sys/ximage/ximagesink.c:
        * sys/xvimage/xvimagesink.c:
          x(v)imagesink: pass the xwindow along to not look at the yet unset var.
          When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.

2009-06-22 11:40:33 +0300  Stefan Kost <ensonic@users.sf.net>

        * sys/ximage/ximagesink.c:
        * sys/ximage/ximagesink.h:
        * sys/xvimage/xvimagesink.c:
        * sys/xvimage/xvimagesink.h:
          x(v)imagesink: catch tags and show title in own window
          Refactor the code that sets the window title. Catch tag-events and use title
          metadata for the window title.

2009-06-21 19:42:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/audiotestsrc/gstaudiotestsrc.c:
          audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
          Also make all the function arrays constant.

2009-06-21 12:27:37 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>

        * gst/audiotestsrc/gstaudiotestsrc.c:
        * gst/audiotestsrc/gstaudiotestsrc.h:
          audiotestsrc: Add support for generating gaussian white noise
          This patch adds support for stationary white Gaussian noise.
          The Box-Muller algorithm is used to generate pairs of independent
          normally-distributed random numbers.
          Fixes bug #586519.

2009-06-20 23:46:28 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst/ffmpegcolorspace/imgconvert.c:
        * gst/ffmpegcolorspace/imgconvert_template.h:
          ffmpegcolorspace: Fix NV12 and NV21 transformations
          Fix some stride problems, fix the nv12 to nv21 direct transformation,
          and implement a direct conversion to yuv444 to save CPU.

2009-06-20 22:36:21 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst/videotestsrc/videotestsrc.c:
          videotestsrc: Fix NV12 painting for odd strides/heights

2009-06-19 22:16:43 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * ext/cdparanoia/gstcdparanoiasrc.c:
          cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
          cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
          Finally fixes #531035.

2009-06-19 21:25:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * ext/cdparanoia/gstcdparanoiasrc.c:
          cdparanoia: try to guess a good cache size if it's set to -1
          Try to guess from the paranoia-mode setting whether playback or
          ripping is wanted, and use a smaller cache size if we're likely
          to be doing playback, to avoid a long startup delay. Since this
          was the value used in older cdparanoia versions, it should be
          fine in any case. See #586331.

2009-06-19 11:27:40 +1000  Jonathan Matthew <jonathan@d14n.org>

        * configure.ac:
        * ext/cdparanoia/gstcdparanoiasrc.c:
        * ext/cdparanoia/gstcdparanoiasrc.h:
          cdparanoia: expose cache size setting
          This setting was added in cdparanoia 10.2.  The default value is good
          for audio extraction, but lower values (previous versions of cdparanoia
          used 150) are better for realtime playback.
          Fixes #586331.

2009-06-19 17:43:03 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

        * gst-plugins-base.spec.in:
          Make build of schro plugin conditional

2009-06-19 15:52:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-sections.txt:
        * gst-libs/gst/rtp/gstbasertppayload.c:
        * gst-libs/gst/rtp/gstbasertppayload.h:
        * win32/common/libgstrtp.def:
          basertppayload: add support for bufferlists
          Based on patch from Ognyan Tonchev.
          See #585559

2009-06-19 15:33:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtp/gstrtpbuffer.c:
          rtpbuffer: use new convenience functions
          New core convenience functions makes the list getters and setters trivial.
          Maybe even too trivial...

2009-06-18 19:07:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * win32/common/libgstrtp.def:
          defs: add new symbol to win32 defs file
          Based on patches by Ognyan Tonchev.
          See #585559

2009-06-18 19:04:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-sections.txt:
        * gst-libs/gst/rtp/gstrtpbuffer.c:
          rtp: cleanups, add _list_get_seq() too
          Clean up the docs a little.
          Add missing _list_get_seq method.
          Add new symbols to the docs

2009-06-18 18:47:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtp/gstrtpbuffer.c:
        * win32/common/libgstrtp.def:
          rtp: cleanups
          Add Since tags to docs
          Move some code around
          Add win32 symbols

2009-06-18 17:46:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtp/gstrtpbuffer.c:
        * gst-libs/gst/rtp/gstrtpbuffer.h:
        * tests/check/libs/rtp.c:
          rtp: add bufferlist support

2009-06-18 18:03:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtp/gstrtpbuffer.c:
          rtp: pass data to macros instead of GstBuffer

2009-06-18 17:42:10 +0100  Jan Schmidt <thaytan@noraisin.net>

        * win32/common/libgstrtsp.def:
          win32: Add gst_rtsp_watch_queue_data() to the exports
          Fix the tests by exporting the new symbol from the win32 dlls

2009-06-18 18:13:22 +0300  Stefan Kost <ensonic@users.sf.net>

        * sys/xvimage/xvimagesink.c:
          xvimagesink: appname might be NULL
          Don't set title if appname is unknown.

2009-06-18 17:58:06 +0300  Stefan Kost <ensonic@users.sf.net>

        * sys/xvimage/xvimagesink.c:
          xvimagesink: set window title from application name

2009-06-09 19:14:00 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspurl.c:
          rtsp: Made the parsing of the RTSP URL scheme more generic.

2009-06-15 13:58:26 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
        * gst-libs/gst/rtsp/gstrtspconnection.h:
          rtsp: Added gst_rtsp_watch_queue_data().
          gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
          but allows for queuing any data block for writing (much like
          gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
          API: gst_rtsp_watch_queue_data()

2009-06-09 16:37:09 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Only extract the session ID from RTSP responses.

2009-06-09 19:06:57 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspurl.c:
          rtsp: Added support for parsing IPv6 addresses in RTSP URLs.

2009-06-09 14:31:18 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Use getaddrinfo() to support both IPv4 and IPv6.

2009-06-17 15:37:53 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Improved base64 decoding in fill_bytes().
          The base64 decoding in fill_bytes() expected the size of the read data to
          be evenly divisible by four (which is true for the base64 encoded data
          itself). This did not, however, take whitespace (especially line breaks)
          into account and would fail the decoding if any whitespace was present.

2009-06-17 14:00:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosrc.c:
          audiosrc: fix get_offset
          When we need to jump to the most recently captured sample, jump to where the
          next sample will be written instead of to some old data.
          Fixes #581460

2009-06-17 13:18:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          audiosink: free the ringbuffer when going to NULL
          Unparent and free the ringbuffer when going to NULL, like we do with the
          audiosrc element. We can do this now because we correctly manage the time
          jumping back to 0.

2009-06-17 13:17:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstaudiosink.c:
        * gst-libs/gst/audio/gstaudiosrc.c:
          audio: correctly handle short read/writes

2009-05-05 15:37:54 +0300  René Stadler <rene.stadler@nokia.com>

        * gst-libs/gst/audio/gstbaseaudiosrc.c:
          baseaudiosrc: add some extra logging for buffer timestamps

2009-06-17 11:22:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/adder/gstadder.c:
          adder: more seeking fixes.
          When a seek failed upstream, make sure the adder sinkpad is set unflushing again
          so that streaming can continue.
          We only have a pending segment when we flushed.
          Set the flush_stop_pending flag inside the appropriate locks and before we
          attempt to perform the upstream seek.
          Add some more comments.
          Use the right lock to protect the flags in flush_stop.
          See #585708

2009-06-17 07:24:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          decodebin2: Free iterator after removing all groups

2009-06-16 19:38:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/video/gstvideofilter.c:
          videofilter: Add a default get_unit_size function
          This returns the correct values for all formats that are handled by
          GstVideoFormat and makes all the custom get_unit_size functions in
          many elements unnecessary.

2009-06-16 18:57:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspdefs.c:
        * gst-libs/gst/rtsp/gstrtspdefs.h:
          rtsp: add Timestamp header field
          fixes #585994

2009-06-16 18:15:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
          playbin2: set smarter target state on uridecodebin
          Set the target state of the newly added uridecodebins to somthing else that
          PAUSED so that we keep their state in sync with the playsink state.
          Fixes #585268

2009-06-16 18:13:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playsink: set the sink flag on the element

2009-06-16 18:09:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          uridecodebin: add debug message

2009-06-16 14:05:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/audio/gstaudiosink.c:
        * gst-libs/gst/audio/gstaudiosrc.c:
          audiosink, audiosrc: do the class_ref()s in the right class_init functions
          Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.

2009-06-15 15:39:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/audio/gstaudiosink.c:
        * gst-libs/gst/audio/gstaudiosrc.c:
          audiosink,audiosrc: ref the audio ring buffer class and type in class_init
          Hack around thread-safety issues in GObject and our racy _get_type()
          functions (we could easily fix the _get_type() functions, but we still
          need to hack around the GObject class races until we require a newer
          GLib version, I think).

2009-06-15 12:57:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosrc.c:
          audiosrc: return FALSE when receiving a SEEK event
          When receiving a seek event, return FALSE as we don't implement seeking.

2009-06-15 11:06:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * tests/examples/seek/seek.c:
          Don't use deprecated GTK API
          Fixes bug #585758.

2009-06-15 11:40:00 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: send flush_stop when seeking failed
          At least do the fix to sent the flush_stop when seeking failed to ensure we
          keep no pads flushing. before it was send when the seeking worked which is just
          plain wrong and was not the intention.

2009-06-12 15:17:14 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Use a more consistent naming of GstRTSPRec variables.

2009-06-12 15:11:05 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
        * gst-libs/gst/rtsp/gstrtspconnection.h:
          rtsp: Call message_sent() callback for all sent messages.
          Previously the messages_sent() callback was only called for messages
          which had a CSeq, which excluded all data messages. Instead of using the
          CSeq as ID, use a simple index counter.

2009-06-14 22:13:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * ext/ogg/gstoggdemux.c:
        * ext/theora/theoradec.c:
        * ext/vorbis/vorbisdec.c:
          oggdemux: post/send tags with the container-format tag
          For this to work properly, theoradec and vorbisdec need to put
          tag events received from upstream into the pending_events list
          so they get pushed out after any newsegment event, not before.

2009-06-14 20:30:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * tests/examples/seek/scrubby.c:
        * tests/examples/seek/seek.c:
        * tests/old/examples/seek/cdplayer.c:
          Don't use deprecated GTK API
          Fixes bug #585758.

2009-06-12 16:31:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/adder/gstadder.c:
          adder: send flush-stop earlier
          When no flush-stop has been sent by upstream, we have to send one ourselves to
          continue playback. Do this as soon as the collect function is called instead of
          after we possibly pushed segment events (that got then flushed out)

2009-06-12 13:55:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/seek.c:
          seek: add shuttle controls

2009-06-12 13:55:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/stepping2.c:
          example: fix compile

2009-06-12 13:52:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/Makefile.am:
          examples: build the stepping2 example

2009-06-12 13:52:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playsink: update for new step API

2009-06-12 13:22:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/ogg/gstoggdemux.c:
          oggdemux: do reverse seeks more accurate
          For reverse seeking with the accurate flag set, try to be more precise by
          seeking a little bit after the requested position.

2009-06-11 22:32:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * ext/ogg/gstogmparse.c:
        * gst/subparse/gstssaparse.c:
        * gst/subparse/gstssaparse.h:
        * gst/subparse/gstsubparse.c:
        * gst/subparse/gstsubparse.h:
          subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
          Make subtitle parsers post a taglist with codec tags, so the application
          knows what kind of subtitle a subtitle stream is. Fixes #576552.

2009-06-11 19:12:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstringbuffer.c:
          ringbuffer: handle border cases in resampler

2009-06-11 13:28:20 +0100  Jan Schmidt <thaytan@noraisin.net>

        * common:
        * docs/libs/Makefile.am:
        * docs/plugins/Makefile.am:
          docs: Update common. Use upload-doc.mak instead of upload.mak

2009-06-11 12:39:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtp/gstbasertppayload.c:
          docs: fix typo

2009-06-11 12:17:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          baseaudiosink: reset accum when dropping samples
          When we are resampling and we drop samples because we paused, reset the accum
          counter because it's now invalid.

2009-06-11 11:16:15 +0100  Jan Schmidt <thaytan@noraisin.net>

        * docs/libs/gst-plugins-base-libs-sections.txt:
        * gst-libs/gst/interfaces/mixer.h:
        * gst-libs/gst/video/gstbasevideodecoder.h:
          docs: Fix a couple of warnings from the docs build.

2009-06-10 21:36:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/audio/testchannels.c:
          Don't include config.h multiple times when build audio testchannel app.
          Fixes build problem on win32 (#585075).

2009-06-10 16:56:51 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst/playback/gstplaybin2.c:
        * gst/playback/gsturidecodebin.c:
          playbin2/uridecodebin: Fix connection-speed propagation
          uridecodebin expects the passed connection-speed value in kbps, so we
          need to divide the value stored in bps by 1000. Also, lower the upper
          limit on the properties to the value that we can actually store in our
          internal guint (which is plenty high enough)

2009-06-10 14:37:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/subparse/gstsubparse.c:
        * tests/check/elements/subparse.c:
          subparse: recognise more subrip timestamp variants
          Be even less restrictive in what we accept for .srt timestamps when
          typefinding and parsing subrip subtitles and add a unit test for
          the 'new' format. Fixes #585197.

2009-06-09 22:00:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtsptransport.h:
          rtsp: add some more docs

2009-06-09 18:24:55 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspmessage.c:
          rtsp: Avoid a compiler warning.

2009-06-09 18:23:28 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspdefs.h:
          rtsp: Updated documentation for GstRTSPResult.
          Moved GST_RTSP_ELAST to be last in the documentation to match the actual
          enum values.

2009-05-20 17:30:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * autogen.sh:
          autogen: remove -Wno-portability from here
          as it is in configure.ac now.

2009-06-09 16:28:20 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Plug a memory leak.
          Free memory related to any partially read and/or written RTSP messages.

2009-06-09 12:09:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          baseaudiosink: no need to cause discont when clipping
          Remove the discont-when-clipping hack now that basesink provides us with
          correctly clipped samples when stepping.

2009-06-08 17:26:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          audiosink: don't align when we clip
          Don't align samples when they were clipped. Not entirely correct but better than
          nothing for now.

2009-06-08 16:41:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/.gitignore:
        * tests/examples/seek/stepping2.c:
          examples: add stepping example in PLAYING
          Add stepping example in PLAYING, audio is a bit distorted because basesink does
          not provide good clipping info yet.

2009-06-08 10:25:00 +0200  Edward Hervey <bilboed@bilboed.com>

        * gst-libs/gst/pbutils/descriptions.c:
          pbutils: Add description for hdv/aux-* formats.

2009-06-07 22:20:33 +0400  LRN <lrn1986@gmail.com>

        * ext/schroedinger/Makefile.am:
          Added libgstbase to schro's LIBADD
          Fixes #585079

2009-06-06 02:15:05 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/tag/gstid3tag.c:
          libgsttag: don't extract genres from empty ID3v1 tags
          If we don't have any other info, don't try to interpret the
          genre field. In particular we don't want to interpret a genre
          of 0 as 'Blues' if no other fields are set and the entire tag
          is just empty.

2009-06-05 18:13:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          decodebin2: make sure varargs are of right type
          Explicitly cast the variables to g_object_set to their right types.

2009-06-05 16:49:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          decodebin2: increase stream probing queues
          When we are probing for streams, we want to set the queue size in such a way
          that we can scan a maximum amount of data without consuming too much memory.
          Therefore, remove the time limit on the queue and only stop scanning after 2MB
          of data.
          See #584104.

2009-06-05 14:06:17 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Fixed a typo.

2009-06-05 14:05:54 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Remove an unused variable.

2009-06-05 13:59:14 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Removed duplicate initialization of conn->writefd.

2009-06-05 13:55:08 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Use #defined status codes.

2009-06-05 13:53:29 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Correct gen_tunnel_reply().
          Prevent gen_tunnel_reply() from generating an incomplete response
          in case an error response code is given.

2009-06-05 10:57:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
        * win32/common/_stdint.h:
        * win32/common/config.h:
        * win32/common/video-enumtypes.c:
          configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
          See #584835. Also update win32 files while we're at it.

2009-06-04 08:57:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
          playbin2: API: Add {audio,video,text}-tags-changed signals
          Fixes bug #584686.

2009-06-03 20:42:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * ext/vorbis/vorbisdec.c:
          vorbisdec: don't put invalid bitrate values into the taglist
          Bitrates are stored as 32-bit signed integers in the vorbis
          identification headers, but seem to be read incorrectly,
          namely as unsigned 32-bit integers, into the vorbis structure
          members which are of type long, which makes our check for
          values <= 0 fail with files that put -1 in there for unset
          values.

2009-06-03 15:52:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/.gitignore:
          ignore: add new stepping app to ignore

2009-06-03 15:31:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/Makefile.am:
        * tests/examples/seek/stepping.c:
          examples: add stepping example.
          Add an example of using playbin2 and frame stepping to simulate variable rate
          playback based on a sine wave.

2009-06-03 12:45:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
        * gst/playback/gstplaysink.h:
          playbin2: also set custom text and subp sinks
          Set the custom subpicture and text sinks along with the custom audio and video
          sinks when needed.
          Fix a little docs blurb too.

2009-06-02 12:10:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
        * gst-libs/gst/rtsp/gstrtspconnection.h:
          rtsp: add G_LIKELY because we can

2009-06-02 09:53:05 +0200  Edward Hervey <bilboed@bilboed.com>

        * gst/typefind/gsttypefindfunctions.c:
          typefindfunctions: Fix caps for ogg typefinder.

2009-05-29 11:10:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-sections.txt:
          docs: remove some cruft from -sections.txt file

2009-06-01 11:31:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
        * tests/examples/seek/seek.c:
          add framestepping to playbin2 and seek

2009-06-01 09:59:22 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: Avoid compiler warnings with -Wextra.

2009-06-01 09:58:27 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.h:
          rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.

2009-06-01 09:43:04 +0200  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/sdp/gstsdpmessage.c:
          sdp: Remove an unused variable.

2009-05-30 14:17:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/ffmpegcolorspace/imgconvert.c:
        * gst/ffmpegcolorspace/imgconvert_template.h:
          ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale

2009-05-29 00:09:15 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst/playback/gstplaybin2.c:
          playbin2: Have playbin recognise PGS subpicture streams
          Recognise PGS subpicture streams and connect them to the SPU pad
          in playsink. Unfortunately this fails badly with negotiation errors
          if the SPU is not recent enough to support the stream. I'm not sure
          how to add format negotiation in yet.

2009-05-21 23:11:29 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst/playback/gstdecodebin2.c:
        * gst/playback/gsturidecodebin.c:
          decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.

2009-05-28 20:37:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2: fix volume handling for audio sinks without "volume" property
          When using an audio sink without a "volume" property, volume control
          would only work for the first song. For the next song, we'd try to
          re-use the existing audio chain, but inadvertently set chain->volume
          to NULL instead of to the existing volume element.

2009-05-28 17:05:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2: cosmetic change to avoid unnecessary line breaks
          Looks nicer and works around gst-indent silliness.

2009-05-28 17:21:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2: don't lose the ref to the volume element
          Only release the ref to the volume element when it is controled by a sink. For
          software volume we never have to fear that it will change.

2009-05-28 15:21:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
        * gst/playback/gstplaysink.c:
          playbin2: actually use configured audio/video sinks
          playbin2 inadvertently used autoaudiosink and autovideosink up to now,
          since it would overwrite the sinks configured via the "audio-sink"
          and "video-sink" properties with the stream-specific group sinks when
          configuring the outputs. Those are usually NULL however, so that would
          overwrite the configured sinks with NULL which makes playbin2 then
          default to the auto sinks. Fix this by keeping a reference to each
          configured sink in playbin2 and setting up the right sinks depending
          on whether there is a stream-specific sink or not.
          Fixes #584020.

2009-05-27 17:37:38 +0300  Stefan Kost <ensonic@users.sf.net>

        * tests/examples/seek/seek.c:
          seek: add volume label and sync with sink volume
          Look at the volume and have the pulsemixer open at same time. Unfortunately
          playbin2 does not emit notify on volume right, so this polls for now.

2009-05-27 18:12:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          decodebin2: remove leftover elements
          Remove all of the elements inside decodebin2 when goint to READY and NULL.
          Makes decodebin2 reusable.
          Fixes #583750

2009-05-27 15:36:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2; release refs to volume/mute properties
          Release the refs to the volume and mute property elemens before setting the
          child elements to READY or NULL.
          Fixes #583318

2009-05-27 12:10:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/gdp/gstgdppay.c:
          gdppay: set caps on outgoing buffers
          Set caps on outgoing buffers because NULL caps confuse basetransform.
          Fixes #583867

2009-05-27 11:08:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/netbuffer/gstnetbuffer.c:
          netbuffer: also note the order of IP4 addresses
          IP4 addresses are also stored in network byte order. Make a note of this in the
          docs.

2009-05-26 22:43:34 +0200  Alessandro Decina <alessandro.d@gmail.com>

        * ext/theora/theoraparse.c:
          theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.

2009-05-26 11:13:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
          This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
          We now require GLib 2.16.

2009-05-26 15:18:09 +0100  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Update common

2009-05-26 15:37:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/netbuffer/gstnetbuffer.c:
          netbuffer: document that the port is network order
          Document the fact that we store the port number in network order in
          GstNetAddress and that the caller should byteswap appropriately.

2009-05-26 15:23:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/gstvideoscale.c:
        * gst/videoscale/vs_4tap.c:
        * gst/videoscale/vs_4tap.h:
        * gst/videoscale/vs_image.c:
        * gst/videoscale/vs_image.h:
        * gst/videoscale/vs_scanline.c:
        * gst/videoscale/vs_scanline.h:
          videoscale: Add support for 16 bit grayscale in native endianness

2009-05-26 14:58:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/ffmpegcolorspace/avcodec.h:
        * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
        * gst/ffmpegcolorspace/imgconvert.c:
          ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian

2009-05-26 14:38:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videotestsrc/videotestsrc.c:
        * gst/videotestsrc/videotestsrc.h:
          videotestsrc: Add support for 16 bit grayscale in native endianness

2009-01-21 12:33:59 +0100  Andy Wingo <wingo@oblong.net>

          add can-activate-pull property to baseaudiosink
          * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
          to baseaudiosink.

2009-05-26 13:14:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/ogg/gstoggdemux.c:
          oggdemux: fix boundary case for seeking.
          When we have exactly 0 bytes left to search, make sure we stop instead of going
          into an infinite loop.

2009-05-26 11:11:03 +0200  Bastien Nocera <hadess at hadess.net>

        * gst-libs/gst/cdda/Makefile.am:
        * gst-libs/gst/cdda/gstcddabasesrc.c:
        * gst-libs/gst/cdda/sha1.c:
        * gst-libs/gst/cdda/sha1.h:
          cddabasesrc: Remove copy of sha1 digest
          Remove our copy of sha1 digest now that we depend on glib 2.16.
          Fixes #536313

2009-05-25 17:54:01 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

        * gst-plugins-base.spec.in:
          Update spec file

2009-05-23 00:33:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/video/gstbasevideodecoder.c:
        * gst-libs/gst/video/gstbasevideoparse.c:
        * gst-libs/gst/video/gstbasevideoutils.c:
        * gst-libs/gst/video/gstbasevideoutils.h:
        * win32/common/libgstvideo.def:
          video: don't expose internal gst_adapter_get_buffer() helper function
          If it's really needed it should go into GstAdapter in core.

2009-05-22 21:29:51 -0700  David Schleef <ds@schleef.org>

        * gst-libs/gst/video/gstbasevideodecoder.c:
          basevideo: Fix memleak

2009-05-22 21:27:58 -0700  David Schleef <ds@schleef.org>

        * ext/schroedinger/gstschrodec.c:
        * ext/schroedinger/gstschroparse.c:
          schro: Fix usage of adapter_masked_scan_uint32
          Because *somebody* changed the API without telling me.

2009-05-22 21:25:06 -0700  David Schleef <ds@schleef.org>

        * ext/schroedinger/gstschro.c:
          schro: Change package name to GST_PACKAGE_NAME

2009-05-22 17:34:10 -0700  David Schleef <ds@schleef.org>

        * gst-libs/gst/video/gstbasevideoencoder.c:
          basevideo: Add preset interface to encoder

2009-05-22 17:31:14 -0700  David Schleef <ds@schleef.org>

        * gst/audioresample/gstaudioresample.c:
          Run liboil benchmark multiple times
          The statistics function requires multiple runs, otherwise
          it causes a divide by zero error.

2009-05-22 19:36:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * m4/gst-fionread.m4:
          m4: fix 'suspicious cache value' warning for gst-fionread.m4
          .. here as well (should really be moved to common, but I'm too lazy).

2009-05-22 17:41:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/vorbis/vorbisdec.c:
          vorbisdec: detect and report errors better
          Check the return values of a couple more libvorbis functions and post an error
          when something is wrong instead of continuing and crashing.

2009-05-22 15:49:14 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstplaysink.c:
          playbin2: fix initial volume and mute handling
          Use two flags to remember volume/mute changes at times when we don't have the
          audiochain yet (e.g. construction). Only set values when they were actualy
          changed. This makes pulseaudio's stream restore functional.

2009-05-22 10:19:51 +0100  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Automatic update of common submodule
          From d3a8fab to 888e0a2

2009-05-22 09:03:22 +0100  Jan Schmidt <thaytan@noraisin.net>

        * win32/common/libgstvideo.def:
          win32: Remove gst_adapter_masked_scan_uint32 from the exports

2009-05-21 10:48:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          audiosink: improve debug message

2009-05-19 18:10:55 -0700  Michael Smith <msmith@songbirdnest.com>

        * gst-libs/gst/tag/gstid3tag.c:
          gstid3tag: Don't extract a track number unless present.
          In ID3v1, a track number is present only if byte 125 is null AND
          byte 126 is non-null. If the track number is not present, don't add
          a track number tag with value 0.

2009-05-20 00:48:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/video/gstbasevideoutils.c:
        * gst-libs/gst/video/gstbasevideoutils.h:
          videoutils: remove adapter methods
          Remove adapter methods now that they are in core.

2009-05-20 00:42:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * win32/common/libgstvideo.def:
          defs: add new symbols

2009-05-19 17:47:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
          autogen: pass -Wno-portability to automake to suppress warnings
          GNU make is needed.

2009-05-19 02:28:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * docs/libs/.gitignore:
          gitignore: remove bogus *.sgml wildcard - these files are tracked in git

2009-05-19 18:41:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/tcp/gsttcpclientsrc.c:
          tcpclientsrc: this is not a live source
          Don't mark us as a live source because we are not.

2009-05-19 18:41:02 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: only send flush_stop when seek failed
          This is still not the ultimate fix. Added some comment to explain the troubles.

2009-05-19 17:17:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          audiosink: return the return value of wait_preroll
          Return the value that _wait_preroll() returned instead of always WRONG_STATE.

2009-05-19 16:45:56 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
        * gst/adder/gstadder.h:
          adder: send flush_stop to match flush_start
          Adder was relying that something else sends a flush stop. When using adder with
          a livesource it was not getting a flush_stop and thus all pads downstream where
          keept flushing. Mark a pending flush_stop and send it when we are working on
          the new segment back in the streaming thread.

2009-05-19 16:02:44 +0300  Stefan Kost <ensonic@users.sf.net>

        * tests/examples/seek/seek.c:
          seek: ui improvements
          Repaint the window black on expose, as this looks nicer when resizing or using
          the expander. Also show time after slider, as this saves a whole line (nice on
          small displays).

2009-04-29 18:36:17 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstdecodebin.c:
          decodebin: use iterators instead of list
          The list api is deprecated. Use threadsafe iterators instead.

2009-05-19 15:35:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          uridecodebin: configure caps on decodebin2
          Implement the caps property by setting the configured caps on new decodebin2
          objects.
          Fixes #582749

2009-05-19 15:34:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          decodebin2: avoid some _caps_ref in some cases
          Only mess with the caps refcount when we configure different caps.

2009-05-19 15:27:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          uridecodebin: fix potential caps leak
          Free the user-configured caps in finalize.

2009-05-19 15:20:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          uridecodebin: add queue after cdda://
          Add a queue2 after the raw output pads of certain sources such as those for uris
          like cdda://
          No tuning of the queue is done yet as the defaults seem to work fine for me.
          Fixes #582528

2009-05-19 12:45:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/ogg/gstoggdemux.c:
          oggdemux: don't loop when at EOS
          When we try to read the last page, don't try to read past the upper boundary, as
          this might cause endless loops.
          See #582942

2009-05-19 11:20:19 +0200  Edward Hervey <bilboed@bilboed.com>

        * gst/audioresample/gstaudioresample.c:
          audioresample: Don't drain remaining buffers after a flush.
          If we were resetted (due to a flush), we can not drain the remaining
          buffers since they would be pushed before a valid new newsegment event.

2009-05-18 22:29:07 -0700  Michael Smith <msmith@syncword.(none)>

        * ext/theora/theoradec.c:
          theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.

2009-05-19 01:13:34 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: add more logging and return value checking

2009-05-19 01:11:45 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: handle the return value from iterator_fold

2009-05-19 01:03:44 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: use the pad in logging as objects
          Helps to differenciate between source and sinks pads.

2009-04-21 22:54:19 +0300  Stefan Kost <ensonic@users.sf.net>

        * tests/examples/seek/seek.c:
          seek: use parser for mp3 and rename variable

2009-05-18 11:08:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/seek.c:
          seek: add playbin2 options in expander
          Add the playbin2 stream selection options inside an expander to preserve some
          space on screen.

2009-02-10 15:29:10 -0800  David Schleef <ds@schleef.org>

        * gst/videotestsrc/videotestsrc.c:
          videotestsrc: Add support for v210 and v216 formats

2009-05-15 16:21:15 -0700  David Schleef <ds@schleef.org>

        * gst-libs/gst/video/gstbasevideocodec.c:
        * gst-libs/gst/video/gstbasevideodecoder.c:
        * gst-libs/gst/video/gstbasevideoencoder.c:
        * gst-libs/gst/video/gstbasevideoparse.c:
          video: remove // comments

2009-05-15 16:18:18 -0700  David Schleef <ds@schleef.org>

        * gst-libs/gst/video/video.c:
        * gst-libs/gst/video/video.h:
          video: Add Y444, v210, v216 formats

2009-05-15 16:12:37 -0700  David Schleef <ds@schleef.org>

        * configure.ac:
        * ext/Makefile.am:
        * ext/schroedinger/Makefile.am:
        * ext/schroedinger/gstschro.c:
        * ext/schroedinger/gstschrodec.c:
        * ext/schroedinger/gstschroenc.c:
        * ext/schroedinger/gstschroparse.c:
        * ext/schroedinger/gstschroutils.c:
        * ext/schroedinger/gstschroutils.h:
          schro: Move schro plugin from Schroedinger
          Previous history is in Schroedinger.  Depends on, and is an example
          of using, GstBaseVideo* base classes.
          Code was reindented, and an #ifdef HAVE_ENCODER removed.

2009-05-15 10:23:08 -0700  David Schleef <ds@schleef.org>

        * gst-libs/gst/video/Makefile.am:
        * gst-libs/gst/video/gstbasevideocodec.c:
        * gst-libs/gst/video/gstbasevideocodec.h:
        * gst-libs/gst/video/gstbasevideodecoder.c:
        * gst-libs/gst/video/gstbasevideodecoder.h:
        * gst-libs/gst/video/gstbasevideoencoder.c:
        * gst-libs/gst/video/gstbasevideoencoder.h:
        * gst-libs/gst/video/gstbasevideoparse.c:
        * gst-libs/gst/video/gstbasevideoparse.h:
        * gst-libs/gst/video/gstbasevideoutils.c:
        * gst-libs/gst/video/gstbasevideoutils.h:
          video: Copy BaseVideo classes from Schroedinger

2009-05-15 23:05:45 +0200  Arnout Vandecappelle <arnout@mind.be>

        * gst/tcp/gstmultifdsink.c:
          multifdsink: add num-fds property
          multifdsink::num-fds

2009-05-15 20:36:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/pbutils/descriptions.c:
          pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000

2009-05-14 11:44:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/vorbis/vorbisenc.c:
          vorbisenc: Implement Preset interface

2009-05-14 11:43:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/theora/theoraenc.c:
          theoraenc: Implement Preset interface

2009-05-14 11:41:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/ogg/gstoggmux.c:
          oggmux: Implement Preset interface

2009-05-14 21:37:22 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst/playback/gstplaysink.c:
          playbin2: Fix cdda:// playback
          Don't send async-start when the playsink has already been configured
          before changing state.

2009-05-14 01:31:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
          configure: require core CVS for gst_adapter_prev_timestamp()
          which is used in the libvisual plugin.

2009-04-22 18:34:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * AUTHORS:
          AUTHORS: fix my email

2009-04-22 18:35:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstaudioclock.c:
          audioclock: make our internal time monotonic
          Make the internal time increase monotonically.

2009-05-13 19:27:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/libvisual/visual.c:
          visual: remove next_ts variable
          We can remove the next_ts variable as we don't use it anymore.

2009-05-13 19:24:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/libvisual/visual.c:
          visual: use new adapter timestamp code
          Use the new adapter timestamp tracking code to make things easier and produce
          vastly better output timestamps.

2009-05-13 01:35:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * po/Makevars:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          po: avoid conflicts of local *.po files with files in git
          Make it so that filenames and line numbers are only stored in the *.pot file
          (which is not in git), but not in the individual *.po files. This information
          is hardly useful for translators in our case, and it should avoid the constant
          conflicts of local *.po files with the ones in git which are caused by the
          source files changing and the line numbers being updated. This commit might
          cause one last merge conflict for you, which you can work around with
          "git checkout po/*.po" before merging or pulling. After that there should
          (hopefully) not be any more local modifications of these files (unless
          someone committed additions or changes to translated strings and the
          *.po files haven't been updated yet, that is).

2009-05-12 23:51:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * tests/check/elements/.gitignore:
        * tests/check/elements/audioresample.c:
          tests: fix audioresample unit test on big endian architectures
          Don't hardcode endianness=1234 in the filtercaps, it will cause
          pad link failures which will result in the test timing out.

2009-05-12 17:18:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/audiotestsrc/gstaudiotestsrc.c:
          audiotestsrc: fix broken enum nick - it should have a hyphen
          The enum nick should be 'sine-table', not 'sine table'. Technically this is
          an API/ABI change I guess, but anyone who was using this and didn't report
          it deserves this.

2009-05-01 01:04:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/audiotestsrc/gstaudiotestsrc.c:
          audiotestsrc: seek to the requested byte offset, not the expected byte offset

2009-05-01 01:03:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/audiotestsrc/gstaudiotestsrc.c:
        * gst/audiotestsrc/gstaudiotestsrc.h:
          audiotestsrc: support more than just one channel

2009-05-12 15:52:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/interfaces/propertyprobe.h:
          propertyprobe: Fix typo in the docs

2009-05-12 12:17:55 +0100  Christian Schaller <christian.schaller@collabora.co.uk>

        * ext/ogg/gstoggmux.c:
        * ext/theora/theora.c:
        * ext/vorbis/vorbis.c:
          Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder

2009-04-30 16:37:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/videorate/gstvideorate.c:
        * gst/videorate/gstvideorate.h:
          videorate: handle invalid timestamps better
          Handle buffers with -1 timestamps better by keeping track of the en time of the
          previous buffer and assuming the -1 timestamp buffer goes right after the
          previous one.
          when we have two buffers that are equally good, output the oldest buffer once to
          minimize latency.
          don't try to calculate latency when the input framerate is unknown.

2009-04-28 11:37:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/ogg/gstoggmux.c:
          oggmux: small debug statement in DISCONT

2009-04-28 11:24:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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>

        * 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.

2009-04-27 10:15:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtp/gstbasertppayload.c:
        * gst-libs/gst/rtp/gstbasertppayload.h:
          rtpdepay: add some more comments

2009-04-17 10:54:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstaudioclock.c:
          audioclock: make sure values are ever increasing

2009-05-05 17:17:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-17 10:57:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-05-12 10:36:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
        * gst/playback/gstplaysink.c:
        * gst/playback/gstplaysink.h:
          playbin2: unify custom sink get/set functions
          Use one function to set/get all of the different sink types.
          cleanup up the subpicture chain too.
          Allow setting a custom subpicture sink.

2009-05-11 18:29:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/interfaces/tunernorm.h:
          interfaces: Seperate some more struct definitions from typedefs

2009-05-11 15:48:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/interfaces/navigation.h:
        * gst-libs/gst/interfaces/videoorientation.h:
        * gst-libs/gst/interfaces/xoverlay.h:
          interfaces: Seperate some more struct definitions from typedefs

2009-05-10 17:28:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * win32/common/libgstinterfaces.def:
          Add new functions to win32 exports

2009-05-10 17:28:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-sections.txt:
          Add new functions to the docs

2009-05-10 17:25:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/interfaces/mixer.c:
        * gst-libs/gst/interfaces/mixer.h:
          interfaces: API: Add gst_mixer_get_mixer_type()
          This is a convenience function that returns the mixer_type
          of the interface struct.

2009-05-10 17:25:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/interfaces/colorbalance.c:
          interfaces: Add docs for gst_color_balance_get_balance_type()

2009-05-10 11:17:19 +0200  Marc-Andre Lureau <marcandre.lureau@gmail.com>

        * autogen.sh:
          Run libtoolize before aclocal
          This unbreaks the build in some cases. Fixes bug #582021

2009-05-07 17:38:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/pango/gsttextrender.c:
          textrender: Correctly initialize the background for ARGB too

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.

2009-05-07 16:52:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/pango/gsttextrender.c:
          textrender: Don't always use ARGB if downstream supports it but take it's preference

2009-05-07 16:48:08 +0200  Kapil Agrawal <kapil@mediamagictechnologies.com>

        * ext/pango/gsttextrender.c:
        * ext/pango/gsttextrender.h:
          textrender: Add support for ARGB and alignment properties
          Fixes bug #581571.

2009-05-07 16:42:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/pango/gsttextrender.c:
          textrender: Add ; after GST_BOILERPLATE to fix indention

2009-05-07 15:10:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/tag/gstvorbistag.c:
          vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists

2009-05-07 14:59:36 +0200  Arnout Vandecappelle <arnout@mind.be>

        * 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.

2009-05-07 14:48:29 +0200  John Millikin <jmillikin@gmail.com>

        * gst-libs/gst/tag/gstvorbistag.c:
          vorbistag: Store cover art in vorbiscomments
          Fixes bug #513373.

2009-05-07 06:14:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/interfaces/colorbalance.c:
        * gst-libs/gst/interfaces/colorbalance.h:
          interfaces: API: Add gst_color_balance_get_balance_type()
          This is a convenience function that returns the balance_type
          of the interface struct.

2009-05-06 17:59:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/interfaces/colorbalance.h:
        * gst-libs/gst/interfaces/colorbalancechannel.h:
        * gst-libs/gst/interfaces/tuner.h:
        * gst-libs/gst/interfaces/tunerchannel.h:
          interfaces: Separate struct definitions from typedefs

2009-05-06 14:03:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * pkgconfig/gstreamer-app-uninstalled.pc.in:
          Fix libdir for uninstalled gstreamer-app library

2009-05-12 01:59:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/pbutils/descriptions.c:
          pbutils: add description for APE tag caps

2009-05-12 01:35:27 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
          configure: bump core requirement to last release
          as that's more likely to be true than that we need
          only 0.21.1.

2009-05-12 01:21:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * common:
        * configure.ac:
          configure: rename CVS -> git in a couple of places

2009-05-12 01:17:53 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
          configure: bump GLib requirement to GLib >= 2.16
          as per the New Regime (see wiki).

2009-05-01 00:09:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-05-07 14:07:44 -0700  Michael Smith <msmith@songbirdnest.com>

        * gst-libs/gst/riff/riff-media.c:
          riff: support UYVY raw 4:2:2 in riff.

2009-05-11 21:20:07 +0100  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
          Back to development -> 0.10.23.1

2009-04-27 22:42:55 -0700  Michael Smith <msmith@syncword.(none)>

        * ext/theora/theoradec.c:
          theoradec: fix buffer overrun on 422 decode.

2009-04-27 21:39:01 -0700  Michael Smith <msmith@syncword.(none)>

        * ext/theora/theoradec.c:
          theoradec: 444 support.

2009-04-27 21:30:04 -0700  Michael Smith <msmith@syncword.(none)>

        * ext/theora/theoradec.c:
          theoradec: handle 422 images (as YUY2).

2009-04-27 21:01:51 -0700  Michael Smith <msmith@syncword.(none)>

        * ext/theora/gsttheoradec.h:
        * ext/theora/theoradec.c:
          theoradec: rearrange code in preparation for 422 and 444 support.

=== release 0.10.23 ===

2009-05-10 23:57:01 +0100  Jan Schmidt <thaytan@noraisin.net>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/gst-plugins-base-plugins.hierarchy:
        * docs/plugins/gst-plugins-base-plugins.interfaces:
        * docs/plugins/gst-plugins-base-plugins.prerequisites:
        * docs/plugins/gst-plugins-base-plugins.signals:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-app.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gdp.xml:
        * docs/plugins/inspect/plugin-gio.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playback.xml:
        * docs/plugins/inspect/plugin-queue2.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-uridecodebin.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * gst-plugins-base.doap:
        * win32/common/_stdint.h:
        * win32/common/config.h:
          Release 0.10.23

2009-05-10 23:56:05 +0100  Jan Schmidt <thaytan@noraisin.net>

        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          Update .po files

2009-05-08 20:32:20 +0100  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
        * win32/common/_stdint.h:
        * win32/common/config.h:
          0.10.22.6 pre-release

2009-05-08 13:09:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2: fix resume after pause
          Don't ignore the state change of the children, they might be doing an ASYNC
          state change.

2009-05-08 11:05:41 +0100  Jan Schmidt <thaytan@noraisin.net>

        * ChangeLog:
        * configure.ac:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          0.10.22.5 pre-release

2009-05-07 22:01:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-05-07 15:19:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-05-06 16:09:52 +0100  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
        * win32/common/_stdint.h:
        * win32/common/config.h:
          0.10.22.4 pre-release

2009-05-06 13:19:34 +0100  Zaheer Merali <zaheerabbas@merali.org>

        * 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.

2009-05-01 16:47:53 +0100  Jan Schmidt <thaytan@noraisin.net>

        * 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

2009-04-29 11:18:42 +0200  Andy Wingo <wingo@oblong.net>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          Revert "add can-activate-pull property to baseaudiosink"
          This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.

2009-04-29 11:18:33 +0200  Andy Wingo <wingo@oblong.net>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          Revert "[baseaudiosink] add docs for can-activate-pull"
          This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.

2009-04-28 18:48:33 +0200  Andy Wingo <wingo@oblong.net>

          [baseaudiosink] add docs for can-activate-pull
          * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
          can-activate-pull.

2009-01-21 12:33:59 +0100  Andy Wingo <wingo@oblong.net>

          add can-activate-pull property to baseaudiosink
          * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
          to baseaudiosink.

2009-04-28 11:32:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-24 18:13:00 +0100  Jan Schmidt <thaytan@noraisin.net>

        * 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.

2009-04-23 08:41:19 +0200  Edward Hervey <bilboed@bilboed.com>

        * 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).

2009-04-21 21:06:59 +0100  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
        * win32/common/_stdint.h:
        * win32/common/config.h:
          0.10.22.3 pre-release

2009-04-21 20:41:23 +0100  René Stadler <mail@renestadler.de>

        * gst/audioresample/gstaudioresample.c:
          audioresample: Fix unused variable in compilation with --disable-gst-debug
          Fixes: #579668

2009-04-21 22:12:28 +0100  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Automatic update of common submodule
          From b3941ea to 6ab11d1

2009-04-21 20:57:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-04-19 18:15:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-04-17 10:34:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtspconnection: don't use GLib-2.16 API, we require only 2.14
          Fixes #579267.

2009-04-17 10:55:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          baseaudiosink: don't unparent the ringbuffer
          when going to NULL, don't unparent the ringbuffer because we don't support going
          back to 0 very well yet.
          Fixes #579203

2009-04-17 10:53:10 +0200  Olivier Crete <tester at tester.ca>

        * 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.

2009-04-16 12:12:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-04-16 12:09:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-04-15 22:59:31 +0100  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
          0.10.22.2 pre-release

2009-04-15 22:56:15 +0100  Jan Schmidt <thaytan@noraisin.net>

        * ChangeLog:
          ChangeLog: regenerate changelog with the gen-changelog script

2009-04-16 00:41:13 +0100  Jan Schmidt <thaytan@noraisin.net>

        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          po: Update po files from TP

2009-04-16 00:40:59 +0100  Jan Schmidt <thaytan@noraisin.net>

        * 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

2009-04-16 00:31:55 +0100  Jan Schmidt <thaytan@noraisin.net>

        * tests/check/libs/video.c:
          check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.

2009-04-16 00:31:00 +0100  Jan Schmidt <thaytan@noraisin.net>

        * 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

2009-04-15 15:35:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/video/video.h:
          video: Fix typo in the docs

2009-04-15 14:53:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/video/video.c:
        * gst-libs/gst/video/video.h:
          video: Add support for YVYU YUV colorspace

2009-04-15 00:17:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-docs.sgml:
        * gst-libs/gst/fft/gstfft.c:
          docs: fix hyperlink and move fft attribution to the right place

2009-04-15 00:02:39 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          log: use G_GUINT64_FORMAT instead of llu

2009-04-14 18:31:52 +0200  Josep Torra <n770galaxy at gmail.com>

        * gst-libs/gst/rtsp/gstrtspdefs.c:
        * gst-libs/gst/rtsp/gstrtspdefs.h:
          RTSP: add missing headers for WMS RTSP
          Add missing headers related to Windows Media RTSP extension.
          Fixes #578942

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

2009-04-14 17:31:31 +0200  Olivier Crete <tester at tester.ca>

        * ext/theora/theoraenc.c:
          theoraenc: factor out keyframe forcing
          See #578656

2009-04-14 17:01:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * AUTHORS:
        * gst-libs/gst/fft/gstfft.c:
          Give credit to Mark Borgerding (kissfft author)
          and add myself to AUTHORS as well. Fixes #575638.

2009-04-14 17:04:06 +0200  Jan Urbanski <j.urbanski at students.mimuw.edu.pl>

        * 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.

2009-04-14 16:53:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-04-14 16:45:20 +0200  Johann Prieur <johann.prieur at gmail.com>

        * 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.

2009-04-14 13:51:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-14 12:10:30 +0100  Hannes Bistry <hannesb@gmx.net>

        * sys/ximage/ximagesink.c:
          ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
          Fixes #570768.

2009-04-14 13:16:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-04-14 13:12:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosink.c:
          baseaudiosink: use new clock time methods
          Use the unadjusted internal clock times to calculate the internal/external
          offset when calibrating the clock.
          When going to NULL, unparent and free the ringbuffer, like we do in the source
          element.
          See #578506

2009-04-14 13:08:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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()

2009-04-14 11:34:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-12 22:26:33 +0200  Luca Ognibene <luca.ognibene at gmail.com>

        * gst/tcp/gstmultifdsink.c:
          multifdsink: fix error in sync-method
          Multifdsink did not handle sync-method=latest-keyframe correctly when the
          soft-limit is set to -1 (unlimited).
          Fixes #578583.

2009-04-10 21:49:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-10 14:12:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-10 13:44:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-10 13:42:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2: fix refcounting of visualisations
          See #577794.

2009-04-10 13:27:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-10 12:27:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/check/elements/appsink.c:
          check: fix appsink test
          Fix the appsink test now that the method signature changed.

2009-04-10 12:26:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
          playbin2: handle missing input-selector
          Gracefully degrade and disable stream selection when input-selector is
          missing.

2009-04-09 23:46:17 +0200  Martin Samuelsson <martin.samuelsson at axis.com>

        * 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.

2009-04-09 18:04:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-08 18:04:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-04-08 16:43:27 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstplaysink.c:
          playbin2: better error message on sink failure
          If we could create the sinks, but the don't work, don't send the missing plugin
          message and report that the state-changed failed.

2009-04-07 22:38:29 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/audio/gstaudiofilter.c:
          audiofilter: don't leak pad-template
          gst_element_class_add_pad_template() does not take ownership.

2009-04-04 21:18:38 +0300  Felipe Contreras <felipe.contreras@gmail.com>

        * common:
          Automatic update of common submodule
          From d0ea89e to b3941ea

2009-04-04 16:28:14 +0200  Edward Hervey <bilboed@bilboed.com>

        * 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.

2009-03-22 09:46:37 +0100  Edward Hervey <bilboed@bilboed.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

2009-04-04 14:53:42 +0200  Edward Hervey <bilboed@bilboed.com>

        * common:
          Automatic update of common submodule
          From f8b3d91 to d0ea89e

2009-04-03 10:51:42 -0700  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstplaybin2.c:
          playbin2: don't leak selector when getting current stream numbers.

2009-04-02 22:28:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-03-31 00:54:30 +0100  Jan Schmidt <thaytan@noraisin.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.

2009-02-04 17:03:07 +0000  Jan Schmidt <thaytan@noraisin.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.

2009-04-02 09:27:07 +0100  Jan Schmidt <thaytan@noraisin.net>

        * tests/check/elements/.gitignore:
          ignores: Ignore the videoscale check binary

2009-04-02 12:13:57 +0100  Jan Schmidt <thaytan@noraisin.net>

        * win32/common/libgstrtsp.def:
          win32: Add gst_rtsp_connection_set_proxy to the win32 exports

2009-04-02 10:42:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * ext/alsa/gstalsamixer.c:
          alsamixer: don't forget to release locks in a few places
          Might fix #576585.

2009-04-02 11:10:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
          videoscale: Don't read over line ends when taking the last Cr or Cb

2009-04-02 10:52:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
          videoscale: Don't write to few pixels and don't mix Cr and Cb
          Fixes bug #577054.

2009-04-01 15:15:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-03-26 19:34:23 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
          docs: add a blurb about redirect messages to playbin2 docs

2009-04-01 09:03:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: fix  little typo in the comments

2009-03-31 17:52:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-03-31 18:13:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-03-31 17:56:24 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspmessage.c:
          docs: fix typo in gst_rtsp_message_unset() API docs

2009-03-31 19:00:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-31 18:57:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspmessage.c:
          Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
          This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.

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.

2009-03-31 12:22:14 +0300  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/rtsp/gstrtspmessage.c:
          rtsp: reset whole message (was sizeof pointer instead of sizeof type)

2009-03-31 00:56:18 +0100  Jan Schmidt <thaytan@noraisin.net>

        * gst-libs/gst/interfaces/mixer.c:
          doc: Fix a typo in the GstMixer docs

2009-03-29 12:01:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_scanline.c:
          videoscale: Fix linear scaling for one byte components
          Fixes bug #577054.

2009-03-29 11:53:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
          videoscale: Fix 4tap scaling of YUYV and friends

2009-03-28 16:08:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

2009-03-28 12:48:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * tests/check/elements/videoscale.c:
          videoscale: Add some more unit tests

2009-03-28 11:51:01 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/gstvideoscale.c:
          videoscale: Use bilinear instead of 4tap scaling for heights < 4
          Partially fixes bug #577054.

2009-03-28 11:45:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

2009-03-28 11:27:56 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_scanline.c:
          videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
          Partially fixes bug #577054.

2009-03-28 10:40:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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).

2009-03-28 10:25:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * tests/check/Makefile.am:
        * tests/check/elements/videoscale.c:
          videoscale: Add a lot of unit tests

2009-03-28 10:06:24 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/gstvideoscale.c:
          videocale: Add support for video/x-raw-gray with bpp=depth=8

2009-03-28 10:01:00 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videotestsrc/videotestsrc.c:
          videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8

2009-03-28 09:43:23 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
          ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format

2009-03-27 19:12:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
          videoscale: Take the next luma value instead of every second next when scaling UYVY and friends

2009-03-27 19:09:47 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/gstvideoscale.c:
          videoscale: Add support for v308 YUV colorspace

2009-03-27 13:15:11 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
          videoscale: Add my copyright to the 4tap scalers

2009-03-27 13:14:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/gstvideoscale.c:
          videoscale: Enable 4-tap scaling for all supported formats

2009-03-27 13:14:00 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
        * gst/videoscale/vs_4tap.h:
          videoscale: Implement 4-tap scaling for RGB565 and RGB555

2009-03-27 10:47:39 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
        * gst/videoscale/vs_4tap.h:
          videoscale: Implement 4-tap scaling for UYVY

2009-03-27 09:33:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
        * gst/videoscale/vs_4tap.h:
          videoscale: Implement 4-tap scaling for YUY2 and YVYU

2009-03-26 22:14:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
        * gst/videoscale/vs_4tap.h:
          videoscale: Implement 4-tap scaling for RGB and BGR

2009-03-26 22:08:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videoscale/vs_4tap.c:
        * gst/videoscale/vs_4tap.h:
          videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats

2009-03-26 11:02:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/pango/gsttextoverlay.c:
          textoverlay: Fix drawing of UYVY text borders

2009-03-26 10:36:27 +0100  Zeeshan Ali <zeeshan.ali@nokia.com>

        * ext/pango/gsttextoverlay.c:
        * ext/pango/gsttextoverlay.h:
          textoverlay: Add support for UYVY colorspace
          Fixes bug #378094.

2009-03-25 19:01:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          decodebin2: do some more cleanup
          Free the groups when we go to READY.
          Allow for NO_PREROLL elements.

2009-03-25 16:37:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-25 16:35:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-25 14:16:25 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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

2009-03-25 11:27:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/audio/gstbaseaudiosrc.c:
          audiosrc: improve 'Dropped n samples' warning message

2009-03-24 19:41:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-24 18:29:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
        * gst/playback/gstplaysink.h:
          playbin2: add more support for subpictures

2009-03-24 17:12:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
        * gst/playback/gstplaysink.c:
        * gst/playback/gstplaysink.h:
          playbin2: first support for subpictures
          Add beginnings of subpicture support.

2009-03-24 15:26:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/seek.c:
          seek: print tags from the different tracks

2009-03-24 12:22:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-24 12:19:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-24 10:42:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2: rescan audio sinks for volume/mute
          Rescan the audio sinks for the mute and volume properties.
          fixes #576180.

2009-03-23 19:40:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-03-23 17:37:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-03-23 16:59:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-23 16:04:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-23 16:19:11 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/tcp/README:
        * gst/tcp/gsttcpclientsink.c:
        * gst/tcp/gsttcpclientsrc.c:
        * gst/tcp/gsttcpserversink.c:
        * gst/tcp/gsttcpserversrc.c:
          docs: add examples for tcp elements, also use correct section name. Fixes #564139
          Updated the examples in the README to actually work. Add them to api docs. Tests
          the api-docs and fix the section names to make the docs actualy show up.
          The example for "tcpserversrc" needs review (might be an element bug).

2009-03-17 09:14:02 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/videoscale/gstvideoscale.c:
          indent: fix damange that gst-indent did some time ago

2009-03-23 15:27:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-23 12:26:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          uridecodebin: reset counter
          reset the number of pending dynamic operations back to 0 when we reuse
          uridecodebin.
          Fixes #576190

2009-03-23 11:38:53 +0100  Edward Hervey <bilboed@bilboed.com>

        * ext/theora/theoradec.c:
          theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
          The problem was that previously we didn't check whether _theora_granule_frame
          returned a negative framecount or not, resulting in bogus timestamps.

2009-03-21 09:46:28 +0100  René Stadler <mail@renestadler.de>

        * ext/vorbis/vorbisenc.c:
          vorbisenc: Set caps on non-header ouput buffers.
          Fixes #576142.

2009-03-20 16:13:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/seek/seek.c:
          seek: Add some more debug
          Add some more info about the selected streams.

2009-03-20 15:47:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-20 14:17:19 +0100  LRN <lrn1986 at gmail 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

2009-03-19 20:31:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-19 19:35:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          decodebin2: more cleanups and docs.
          Add some more comments and use g_list_prepend().

2009-03-19 19:19:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-19 19:03:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-18 10:45:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/icles/playbin-text.c:
          icles: print better error and warning messages
          --

2009-03-17 22:53:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

2009-03-17 19:53:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-17 19:09:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-17 15:39:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-17 11:54:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-17 11:51:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-17 11:30:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/icles/playbin-text.c:
          example: use appsink instead of fakesink
          Use appsink instead of fakesink to get the subtitles.
          Make things more pretty.

2009-03-17 11:24:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-17 11:03:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/check/elements/appsink.c:
          tests: fix include in the appsink test
          Fix dist by doing the right include.

2009-03-16 16:42:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-16 16:16:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-16 15:31:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/examples/app/appsrc-ra.c:
        * tests/examples/app/appsrc-seekable.c:
        * tests/examples/app/appsrc-stream.c:
        * tests/examples/app/appsrc-stream2.c:
          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).

2009-03-15 19:57:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-03-11 12:59:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-03-13 15:22:42 +0000  Jan Schmidt <thaytan@noraisin.net>

        * tests/check/Makefile.am:
          check: Ignore alsamixer in the states test too

2009-03-13 15:22:11 +0000  Jan Schmidt <thaytan@noraisin.net>

        * sys/v4l/v4l_calls.c:
          v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.

2009-03-13 16:19:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-13 15:29:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-13 15:27:51 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-13 15:26:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/ogg/gstoggdemux.c:
          oggdemux: add some debug info
          Add some debug info to log when the seek worked.

2009-03-13 15:14:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-13 11:49:10 +0000  Jan Schmidt <thaytan@noraisin.net>

        * ext/vorbis/vorbisdec.c:
          vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.

2009-03-13 11:48:28 +0000  Jan Schmidt <thaytan@noraisin.net>

        * ext/theora/theoradec.c:
          theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.

2009-03-12 18:27:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gsturidecodebin.c:
          playbin2: fix raw elements like cdda://
          Fix a fixme with a one liner and make cd playback work again.

2009-03-12 17:47:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-24 15:58:42 +0000  Jan Schmidt <thaytan@noraisin.net>

        * gst-libs/gst/tag/gstvorbistag.c:
          vorbistag: Protect memory allocation calculation from overflow.
          Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586

2009-03-12 11:34:20 +0000  Jan Urbanski <jurbanski@flumotion.com>

        * gst-plugins-base.spec.in:
          Spec: fix up deps

2009-03-11 18:45:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: fix parsing of the timeout parameter
          --

2009-03-11 16:20:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspmessage.c:
          rtsp: fix g_return condition
          when parsing a data message, we require a data message.

2009-03-11 13:33:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-03-11 14:08:10 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-11 14:07:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-10 21:00:26 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: log details in getcaps like in setcaps

2009-03-10 13:11:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * win32/MANIFEST:
          win32: update MANIFEST, fixing 'make dist'

2009-03-09 23:12:00 +0000  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Automatic update of common submodule
          From 7032163 to f8b3d91

2009-03-09 16:19:40 +0100  Jonathan Matthew <notverysmart at gmail dot com>

        * gst/typefind/gsttypefindfunctions.c:
          typefind: add photoshop typefind functions
          Add photoshop typefind functions.
          Fixes #574516.

2009-03-09 15:46:21 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-09 13:53:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-09 13:53:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspmessage.c:
          rtsp: use gstreamer dump mem
          --

2009-03-09 13:51:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: use glib base64 encoder
          --

2009-03-06 19:28:37 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
          Unblock blocked ghostpads when shutting down.  Fixes #574293.

2009-03-09 10:03:13 +0100  Edward Hervey <bilboed@bilboed.com>

        * gst-libs/gst/riff/riff-media.c:
          Riff: Add mapping for Fraps video codec.
          Found through insanity testrun. Confirmed mapping in libavformat.

2009-03-09 09:07:13 +0100  Edward Hervey <bilboed@bilboed.com>

        * 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.

2009-03-09 09:06:40 +0100  Edward Hervey <bilboed@bilboed.com>

        * gst/typefind/gsttypefindfunctions.c:
          typefind: Use the proper data pointer instead of poking random memory.

2009-03-08 18:17:48 +0100  LRN <lrn1986@gmail.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: fix compilation on windows.
          Remove unused variable when building for windows.
          Fixes #574443.

2009-03-08 12:03:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From ffa738d to 7032163

2009-03-08 11:19:00 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From 3f13e4e to ffa738d

2009-03-07 11:44:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From 3c7456b to 3f13e4e

2009-03-07 10:44:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * common:
          Automatic update of common submodule
          From 57c83f2 to 3c7456b

2009-03-06 19:02:58 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-06 18:53:17 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/riff/riff-media.c:
          riff: add theora mapping
          Add theora mappings. See #574169.

2009-03-06 16:31:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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()

2009-03-06 10:35:01 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * Makefile.am:
        * win32/common/audio-enumtypes.c:
          win32: indent copied *-enumtypes.c files in make win32-update

2009-03-06 10:35:56 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * win32/MANIFEST:
          win32: update MANIFEST

2009-03-06 10:30:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * configure.ac:
        * win32/common/config.h:
          win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define

2009-03-06 10:05:11 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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).

2009-03-06 10:03:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * .gitignore:
          gitignore: ignore more

2009-03-06 10:37:38 +0100  Julien Moutte <julien@fluendo.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          Fix build on Mac OS X

2009-03-05 15:42:23 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstdecodebin2.c:
          decodebin2: don't stay connected to notify::caps after negotiation
          Disconnect the notify::caps signal in our callback (it'll be re-added
          if we're not, in fact, finished getting complete caps). Ensures that
          caps changes mid-stream (e.g. from an mp3 that changes from
          stereo->mono mid-file) don't cause us to try to add a new pad.

2009-03-05 13:48:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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>

        * docs/plugins/gst-plugins-base-plugins-sections.txt:
          docs: add appsrc symbols to standard section
          --

2009-03-05 12:27:16 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/adder/gstadder.c:
          adder: add variants for unsigned to fix warnings for unneeded check
          For unsigned int out+in can't be < 0.

2009-03-05 10:58:12 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/subparse/gstsubparse.c:
          subparse: use the right variable in debug log, encoding is not yet initialized

2009-03-05 10:51:25 +0200  Stefan Kost <ensonic@users.sf.net>

        * sys/v4l/v4l_calls.c:
          v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix

2009-03-05 10:39:33 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/audioresample/gstaudioresample.c:
          audioresample: add missing break in event handling, remove dead code

2009-03-04 16:24:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-04 16:11:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-04 12:21:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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>

        * 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.

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.

2009-02-28 15:23:20 -0800  LRN <lrn1986@gmail.com>

        * gst/subparse/gstsubparse.c:
          subparse: Convert regex code to GRegex code
          Fixes: #572993.  Patch author prefers to use an alias, contact
          ds if you actually need a real name.
          Signed-off-by: David Schleef <ds@schleef.org>

2009-03-02 16:13:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtsp: remove debugging g_message
          --

2009-03-02 16:03:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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()

2009-03-02 15:48:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-03-02 12:48:18 +0100  Edward Hervey <bilboed@bilboed.com>

        * win32/common/libgstrtsp.def:
          win32: Add gst_rtsp_connection_get_url definition
          No, I'm not wim's buildslave, seriously.

2009-03-02 10:58:49 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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()

2009-03-01 18:31:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-02-28 11:07:04 -0800  David Schleef <ds@schleef.org>

        * configure.ac:
          Bump glib requirement to 2.14

2009-02-28 19:37:53 +0100  LRN <lrn1986@gmail.com>

        * ext/gio/gstgiobasesink.c:
          gio: Use correct format modifier for size_t
          Fixes bug #573528.

2009-02-28 19:35:33 +0100  LRN <lrn1986@gmail.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtspconnection: Use correct types for some functions on Win32
          Fixes bug #573529.

2009-02-28 13:11:59 +0100  Edward Hervey <bilboed@bilboed.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtspconnection: Fix warning about using unitialized value.

2009-02-28 12:41:28 +0100  Edward Hervey <bilboed@bilboed.com>

        * 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

2009-02-27 11:14:25 +0200  Stefan Kost <ensonic@users.sf.net>

        * ext/alsa/gstalsa.c:
          alsa: release pcminfo after the strdup

2009-02-26 17:38:47 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/rtsp/gstrtsprange.c:
          rtsprange: don't leak the range in case of parsing error.
          Free the gstRTSPTimeRange if we don't return it. Also simplify
          gst_rtsp_range_free() as it is valid to pass NULL to g_free().

2009-02-26 16:47:39 +0200  Stefan Kost <ensonic@users.sf.net>

        * ext/alsa/gstalsa.c:
          alsa: cleanup name lookup.
          We can break, once we have a name to make sure, we won't read it ever twice.

2009-02-26 16:09:03 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/subparse/gstsubparse.c:
          subparse: don't leak line, if flushing

2009-02-26 16:03:39 +0200  Stefan Kost <ensonic@users.sf.net>

        * ext/gio/gstgiosink.c:
          giosink: reflow error handling to not leak uri

2009-02-26 15:53:10 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
        * gst/ffmpegcolorspace/imgconvert.c:
          ffmpegcolorspace: remove unused code/variables

2009-02-26 12:10:47 +0200  Stefan Kost <ensonic@users.sf.net>

        * sys/ximage/ximagesink.c:
          ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps

2009-02-26 16:44:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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>

        * 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.

2009-02-26 09:52:59 +0100  Edward Hervey <bilboed@bilboed.com>

        * win32/common/libgstapp.def:
          win32: Add new libgstapp symbol

2009-02-26 10:07:21 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-02-26 10:06:23 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstplaybasebin.c:
          docs: fix random text after since: tag. Also fix class name to make the docs actual appear.

2009-02-26 09:56:16 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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

2009-02-26 09:53:03 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/video/video.h:
          docs: fix newly added interlace constants and plug holes in video format docs

2009-02-26 09:35:43 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst-libs/gst/app/gstappsink.c:
        * gst-libs/gst/app/gstappsrc.c:
        * gst-libs/gst/audio/gstaudiofilter.c:
        * gst-libs/gst/audio/gstringbuffer.c:
        * gst-libs/gst/rtp/gstrtcpbuffer.c:
          docs: don't put random stuff in tags.
          Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
          tag to append text again to the documentation body.

2009-02-06 11:10:15 +0200  Stefan Kost <ensonic@users.sf.net>

        * sys/ximage/ximagesink.c:
          ximagsink: do not access uninitialized height variable.
          Exit like in xvimagesink, if we have partial caps.

2009-02-25 20:26:05 -0800  David Schleef <ds@schleef.org>

        * 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.

2009-02-25 16:25:33 +0100  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          Fixed a typo.

2009-02-25 11:31:02 +0100  Peter Kjellerstedt <pkj@axis.com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
        * gst/tcp/gstmultifdsink.c:
          rtsp, multifdsink: Unify the use of union gst_sockaddr.

2009-02-25 14:22:35 +0000  Jan Schmidt <thaytan@noraisin.net>

        * common:
        * configure.ac:
          build: Update shave init statement for changes in common. Bump common.

2009-02-25 13:16:32 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * sys/xvimage/xvimagesink.c:
        * sys/xvimage/xvimagesink.h:
          xvimageink: protect buffer_alloc from shutdown
          Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
          crashes when the sink is shutdown.

2009-02-25 12:43:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-25 12:42:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-25 11:31:52 +0000  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Automatic update of common submodule
          From 9cf8c9b to a6ce5c6

2009-02-25 09:52:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * gst-libs/gst/riff/riff-media.c:
          riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
          Fixes: #565777

2009-02-25 12:07:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-25 11:07:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * tests/icles/stress-playbin.c:
          Print the errors more clearly
          Print some more verbose messages when dealing with errors.

2009-02-25 10:08:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-25 10:05:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-25 08:22:00 +0100  Edward Hervey <bilboed@bilboed.com>

        * gst/playback/gstplaybin2.c:
          Playbin2: Don't leave unused parameters in debug statements.
          Fixes build on macosx

2009-02-24 10:33:05 +0100  Edward Hervey <bilboed@bilboed.com>

        * gst-libs/gst/riff/riff-media.c:
          Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)

2009-02-24 18:43:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
          Add some G_UNLIKELY because we can
          Add a G_UNLIKELY when checking the shutdown variable.

2009-02-24 17:23:58 +0000  Garret D'Amore <garrett.damore@sun.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

2009-02-24 17:03:08 +0000  Jan Schmidt <thaytan@noraisin.net>

        * gst/tcp/gstmultifdsink.c:
          multifdsink: Fix strict aliasing error using a union

2009-02-24 16:49:40 +0000  Jan Schmidt <thaytan@noraisin.net>

        * 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.

2009-02-24 16:08:49 +0000  Jan Schmidt <thaytan@noraisin.net>

        * 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.

2009-02-24 14:36:39 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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

2009-02-24 14:06:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/ffmpegcolorspace/avcodec.h:
        * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
        * gst/ffmpegcolorspace/imgconvert.c:
          ffmpegcolorspace: Add conversion from/to YVYU colorspace
          Fixes bug #572872.

2009-02-24 13:42:01 +0100  Jonas Danielsson <jonas.danielsson@axis.com>

        * 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.

2009-02-19 17:16:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

        * gst/playback/gstplaybin2.c:
          playbin2: fix deadlock when shutting down.  Fixes #572577.

2009-02-19 17:15:18 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

        * tests/icles/stress-playbin.c:
          stress-playbin: make more flexible, e.g. also useful for playbin2

2009-02-24 12:11:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          Match WSAStartup and WSACleanup correctly
          Don't randomly call WSAStartup and WSACleanup but instead call the startup when
          we create a connection and cleanup when we free it again. Because the internal
          datastructure is refcounted, this should not cause any refcounting leaks when
          the connection is managed correctly.
          Fixes #562794.

2009-02-18 11:59:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

        * gst/playback/gstplaysink.c:
          playbin2/playsink: Set audiotee to PAUSED state in all cases.  Fixes #565105.

2009-02-23 10:57:42 -0800  David Flynn <davidf@rd.bbc.co.uk>

        * pkgconfig/gstreamer-app-uninstalled.pc.in:
        * pkgconfig/gstreamer-audio-uninstalled.pc.in:
        * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
        * pkgconfig/gstreamer-fft-uninstalled.pc.in:
        * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
        * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
        * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
        * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
        * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
        * pkgconfig/gstreamer-riff-uninstalled.pc.in:
        * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
        * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
        * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
        * pkgconfig/gstreamer-tag-uninstalled.pc.in:
        * pkgconfig/gstreamer-video-uninstalled.pc.in:
          Add srcdir to includes for out-of-source builds
          When you use gstreamer uninstalled and build outside
          the source tree, the includes need to be specified for
          both the source tree and the build tree.
          Signed-off-by: David Schleef <ds@schleef.org>

2009-02-22 17:23:52 +0000  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
        * docs/libs/Makefile.am:
        * docs/plugins/Makefile.am:
          Use shave for the build output

2009-02-23 12:17:07 +0100  Edward Hervey <bilboed@bilboed.com>

        * win32/common/libgstrtsp.def:
          win32: Add new symbol to libgstrtsp.def

2009-02-23 10:57:08 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-22 19:20:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * tests/check/libs/netbuffer.c:
          Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)

2009-02-22 19:19:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * ext/theora/theoraparse.c:
          theoraparse: Use the correct unref functions

2009-02-22 19:18:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * sys/ximage/ximagesink.c:
        * sys/xvimage/xvimagesink.c:
          x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()

2009-02-22 19:12:00 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst-libs/gst/tag/gsttagdemux.c:
          tagdemux: Unref the actual buffer instead of the memory address of the buffer

2009-02-22 15:47:53 +0000  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Automatic update of common submodule
          From 5d7c9cc to 9cf8c9b

2009-02-22 14:49:29 +0100  Edward Hervey <bilboed@bilboed.com>

        * win32/common/libgstrtsp.def:
        * win32/common/libgstvideo.def:
          win32/common: Update .def files for recent API addition

2009-02-22 13:43:35 +0100  Edward Hervey <bilboed@bilboed.com>

        * tests/check/libs/rtp.c:
          tests: Fix indentation

2009-02-22 13:42:33 +0100  Edward Hervey <bilboed@bilboed.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.

2009-02-21 11:13:36 -0800  David Schleef <ds@schleef.org>

        * common:
          Automatic update of common submodule
          From 80c627d to 5d7c9cc

2009-02-20 17:26:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-20 12:35:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-20 09:50:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-19 12:18:29 -0800  Benjamin M. Schwartz <bens@alum.mit.edu>

        * ext/theora/gsttheoraenc.h:
        * ext/theora/theoraenc.c:
          theoraenc: Add property for speed level control
          Add property "speed-level" to control the amount of motion searching
          the encoder does.  This is only available in libtheora >= 1.0 and
          will silently fail with earlier libraries.  Fixes: #572275.
          Signed-off-by: David Schleef <ds@schleef.org>

2009-02-19 17:40:45 +0100  Edward Hervey <bilboed@bilboed.com>

        * 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>

        * 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).

2009-02-19 15:51:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-26 10:31:14 +0100  Edward Hervey <bilboed@bilboed.com>

        * gst-libs/gst/riff/riff-media.c:
          Add some more mappings for h264 in riff

2009-02-19 10:49:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * win32/common/libgstrtsp.def:
          Add new RTSP symbols to def files
          Add the new RTSP symbols to the windows def file.

2009-02-19 10:44:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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>

        * 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()

2009-02-18 17:42:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * docs/libs/gst-plugins-base-libs-sections.txt:
        * gst-libs/gst/rtsp/gstrtspconnection.c:
        * gst-libs/gst/rtsp/gstrtspconnection.h:
          Add RTSP channel object for async io
          Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
          that the connection can be monitored from a maincontext. This allows us to
          operate in ASYNC mode, which is handy when building a server.
          Rework the old code to use the async code under the hood.
          API: gst_rtsp_channel_new()
          API: gst_rtsp_channel_unref()
          API: gst_rtsp_channel_attach()
          API: gst_rtsp_channel_queue_message()

2009-02-15 07:30:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/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.

2009-02-13 10:10:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
        * gst/videotestsrc/videotestsrc.c:
          ffmpegcolorspace/videotestsrc: Use v308 instead of V308

2009-02-12 19:02:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * 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.

2009-02-12 18:17:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videotestsrc/videotestsrc.c:
          videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
          Partially fixes bug #571147.

2009-02-12 09:18:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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).

2009-02-11 16:39:55 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-02-11 16:39:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * .gitignore:
          Make git ignore backup files.

2009-02-10 20:38:58 -0800  Michael Smith <msmith@syncword.(none)>

        * 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.

2009-02-10 17:16:07 -0800  Michael Smith <msmith@songbirdnest.com>

        * 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.

2009-02-10 17:10:33 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstplaysink.c:
          playbin2: Look for volume/mute properties recursively in audio element.
          Rather than only checking for volume property on the audio sink
          directly, recursively look for it on sinks within it (if it's a bin).
          Allows use of sink-as-volume-control where the application has supplied
          an audio-sink bin that includes a real audio sink internally.

2009-02-10 18:29:22 +0000  Christian Schaller <cschalle@crazyhorse.localdomain>

        * gst-plugins-base.spec.in:
          Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base

2009-02-10 17:39:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

        * gst/videotestsrc/videotestsrc.c:
          videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
          Partially fixes bug #571147.

2009-02-10 17:37:06 +0100  Peter Kjellerstedt <pkj@axis.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().

2009-02-10 11:00:12 +0100  Antoine Tremblay <hexa00@gmail.com>

        * ext/alsa/gstalsamixer.c:
          alsamixer: Fix race condition that made alsamixer not working properly
          This is due to race conditions between functions that
          modified the mixer like set_volume and
          snd_mixer_handle_events since the handle_events
          can now be called at any time.
          Fixed by adding locking around any snd_mixer call
          since even read functions can modify the mixer stucture, since
          alsa likes to clear it's values before reading new ones.
          The favorite race condition seemed to be that set_volume
          called read_elem (in alsalib) that reset the volumes to
          0 and then read them with read_x_volume. This read looped
          on each channel and as the race condition occured the
          channels value could be anything , most of the time
          it was 0. Thus no value was read or only the value of
          one channel was and the volume was reset to 0.
          Fixes bug #478512.

2009-02-09 12:02:21 +0100  Edward Hervey <bilboed@bilboed.com>

        * common:
          Bump revision to use for common submodule.

2009-02-05 15:47:00 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-02-04 17:03:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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()

2009-02-04 13:16:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-04 13:53:30 +0200  Stefan Kost <ensonic@users.sf.net>

        * gst/playback/gstplaysink.c:
          playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
          The flags where present but actually not been taken into account.

2009-02-04 12:06:38 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-02-04 11:18:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-04 10:31:21 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-02-03 15:26:08 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-02-03 10:52:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * win32/common/libgstrtsp.def:
          Add new methods
          Add new methods to the windows def file.

2009-02-02 17:25:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

        * 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.

2009-02-02 18:05:42 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-02-02 15:43:03 +0200  Stefan Kost <ensonic@users.sf.net>

        * 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.

2009-02-02 13:30:42 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-02-02 12:54:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * gst/playback/gstdecodebin2.c:
        * gst/playback/gsturidecodebin.c:
          Fix documentation for autoplug-select
          fix the documentation strings for the autoplug-select signal.
          Fixes #570142.

2009-02-02 10:09:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-31 18:45:47 +0200  Stefan Kost <ensonic@users.sf.net>

        * docs/libs/gst-plugins-base-libs-docs.sgml:
          Its "Base Library" and not just "Library".

2009-01-31 18:44:32 +0200  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.

2009-01-30 17:48:23 -0800  Michael Smith <msmith@songbirdnest.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.

2009-01-30 17:47:07 -0800  Michael Smith <msmith@songbirdnest.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.

2009-01-30 15:23:23 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstplaybin2.c:
          Unref source in playbin2's finalize method

2009-01-30 12:04:01 -0800  Michael Smith <msmith@songbirdnest.com>

        * 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.

2009-01-30 11:04:37 -0800  Michael Smith <msmith@songbirdnest.com>

        * 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.

2009-01-30 10:54:12 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstplaybin2.c:
          Unref peer request pad after releasing it, since we hold a reference.

2009-01-30 10:52:52 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstplaybin2.c:
          Fix caps leak in playbin2.

2009-01-30 10:51:11 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstplaybin2.c:
          Unref active pad from selector when finding active stream.

2009-01-30 10:49:55 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gstplaybin2.c:
          Free uris when finalizing playbin2 instance.

2009-01-30 10:38:17 -0800  Michael Smith <msmith@songbirdnest.com>

        * gst/playback/gsturidecodebin.c:
          Unref pads when iterating over them in analyse_source.
          Fixes leak of source's srcpad when using uridecodebin.

2009-01-30 22:22:07 +0200  Stefan Kost <ensonic@users.sf.net>

        * docs/plugins/gst-plugins-base-plugins-docs.sgml:
          Add releaseinfo with online url.

2009-01-30 17:58:15 +0000  Jan Schmidt <jan.schmidt@sun.com>

        * gst/playback/gstplaybasebin.c:
          Fix compilation warning on Forte

2009-01-30 17:16:39 +0000  Jan Schmidt <jan.schmidt@sun.com>

        * gst/adder/gstadder.c:
          Don't do void pointer arithmetic.

2009-01-30 17:25:51 +0000  Jan Schmidt <thaytan@noraisin.net>

        * common:
          Bump common

2009-01-30 08:50:53 +0100  Edward Hervey <bilboed@bilboed.com>

        * autogen.sh:
        * common:
          Use a symbolic link for the pre-commit client-side hook

2009-01-30 08:12:42 +0100  Edward Hervey <bilboed@bilboed.com>

        * .gitignore:
          Add more files/directories to ignore

2009-01-29 14:00:30 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-29 11:55:10 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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()

2009-01-29 11:51:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-28 11:48:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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()

2009-01-26 17:59:37 -0800  Michael Smith <msmith@songbirdnest.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).

2009-01-26 23:32:09 +0000  Jan Schmidt <thaytan@noraisin.net>

        * ext/gio/Makefile.am:
          Fix include order for gio plugin

2009-01-23 23:59:48 +0000  Jan Schmidt <thaytan@noraisin.net>

        * win32/common/config.h:
          Update win32 config.h for 0.10.22.1 dev cycle

2009-01-23 23:16:11 +0000  Jan Schmidt <thaytan@noraisin.net>

        * .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

2009-01-23 12:31:06 +0100  Sebastian Dröge <sebastian.droege@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-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.

2009-01-23 11:44:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-23 11:40:26 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-23 11:37:45 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

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:
          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

2009-01-23 11:27:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-23 11:23:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * ext/alsa/gstalsasink.c:
          Improve debug message
          Improve the debug message when alsa returns an error.

2009-01-23 11:07:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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

2009-01-23 10:19:27 +0100  Benjamin Gaignard <benjamin@gaignard.net>

        * gst/typefind/gsttypefindfunctions.c:
          Add typefinder for Mobile XMF. Fixes bug #568707.

2009-01-23 10:00:11 +0100  Brian Cameron <brian.cameron@sun.com>

        * 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.

2009-01-22 22:09:47 +0000  Jan Schmidt <thaytan@noraisin.net>

        * gst/playback/gstplaybasebin.c:
          Fix use-after-unref problem noticed by Josep Torra Valles, and run
          gst-indent

2009-01-22 17:46:59 +0200  Stefan Kost <ensonic@users.sf.net>

        * common:
          Update common snapshot.

2009-01-22 13:47:24 +0100  Sebastian Dröge <slomo@circular-chaos.org>

        * common:
          Fix pre-commit hook

2009-01-22 13:12:02 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

          Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base

2009-01-22 10:14:28 +0100  Sebastian Dröge <slomo@circular-chaos.org>

        * 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.

2009-01-22 11:32:56 +0000  Jan Schmidt <thaytan@noraisin.net>

        * configure.ac:
          Back to devel -> 0.10.22.1

2009-01-22 05:57:53 +0100  Edward Hervey <bilboed@bilboed.com>

        * autogen.sh:
        * common:
          Install and use pre-commit indentation hook from common

2009-01-21 13:09:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

        * 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.

2009-01-21 04:31:32 +0100  Edward Hervey <bilboed@bilboed.com>

        * autogen.sh:
          autogen.sh : Use git submodule

=== release 0.10.22 ===

2009-01-19 23:10:50 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.hierarchy:
        * docs/plugins/gst-plugins-base-plugins.interfaces:
        * docs/plugins/gst-plugins-base-plugins.prerequisites:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-app.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gdp.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playback.xml:
        * docs/plugins/inspect/plugin-queue2.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-uridecodebin.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * gst-plugins-base.doap:
        * po/LINGUAS:
        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/en_GB.po:
        * po/es.po:
        * po/fi.po:
        * po/fr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
        * win32/common/config.h:
          Release 0.10.22
          Original commit message from CVS:
          Release 0.10.22

2009-01-19 22:01:01 +0000  Jan Schmidt <thaytan@mad.scientist.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/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.

2008-12-13 16:21:12 +0000  Edward Hervey <bilboed@bilboed.com>

          m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
          Original commit message from CVS:
          * m4/Makefile.am:
          And a couple more .m4 that don't exist anymore with gettext 0.17

2008-12-13 12:41:56 +0000  Edward Hervey <bilboed@bilboed.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.

2008-12-13 06:57:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_slave_method_get_type),
          (gst_base_audio_sink_class_init):
          * gst-libs/gst/audio/gstbaseaudiosink.h:
          * gst-libs/gst/audio/gstbaseaudiosrc.c:
          (gst_base_audio_src_slave_method_get_type),
          (gst_base_audio_src_class_init):
          * gst-libs/gst/audio/gstbaseaudiosrc.h:
          API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
          public API. This is needed for the C++ bindings to be able
          to use this base classes. Fixes bug #564200, #564206.

2008-12-12 19:41:28 +0000  Edward Hervey <bilboed@bilboed.com>

          gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
          Original commit message from CVS:
          * gst-libs/gst/cdda/gstcddabasesrc.c:
          (gst_cdda_base_src_handle_event):
          Remove erroneous gst_buffer_ref().
          * tests/check/libs/rtp.c: (GST_START_TEST):
          Don't forget to unref the buffer once you're done with it.

2008-12-12 13:06:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst/playback/: XRef to GstXOverlay.
          Original commit message from CVS:
          * gst/playback/gstplaybin.c:
          * gst/playback/gstplaybin2.c:
          XRef to GstXOverlay.

2008-12-12 10:54:45 +0000  Edward Hervey <bilboed@bilboed.com>

          gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
          Original commit message from CVS:
          * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
          Free the factory array when finalizing.
          * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
          Use a GstStaticPadTemplate since the src pad caps are fixed.

2008-12-12 07:17:21 +0000  Edward Hervey <bilboed@bilboed.com>

          ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
          Original commit message from CVS:
          * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
          (gst_vorbis_enc_init):
          Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
          pad templates.

2008-12-12 07:15:22 +0000  Edward Hervey <bilboed@bilboed.com>

          gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
          (gst_riff_create_video_template_caps):
          Add mapping for VP6 in avi/riff.

2008-12-11 15:49:12 +0000  Edward Hervey <bilboed@bilboed.com>

          gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
          Original commit message from CVS:
          * gst/subparse/samiparse.c: (sami_context_push_state),
          (sami_context_pop_state), (start_sami_element), (end_sami_element):
          Some versions of libxml seem to be very picky as to strict formatting
          of the input and never 'close' the final </body> tag.
          In order to fix that bad behaviour, we trigger the flushing of
          remaining data on both </body> and </sami>.
          Fixes #557365

2008-12-11 12:32:03 +0000  Guillaume Emont <guillaume@fluendo.com>

          gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
          Original commit message from CVS:
          Patch by: Guillaume Emont <guillaume at fluendo dot com>
          * gst/typefind/gsttypefindfunctions.c: (plugin_init):
          Add typefinders for MS Word files and OS X .DS_Store files to
          prevent them to be recognized as MPEG files. Fixes bug #564098.

2008-12-11 11:04:14 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaysink.c: Add some more debug info.
          Original commit message from CVS:
          * gst/playback/gstplaysink.c: (gen_audio_chain),
          (gst_play_sink_reconfigure):
          Add some more debug info.
          Fix linking of just an encoded sink.
          Handle failure to create a sink chain more gracefully than crashing.

2008-12-11 10:33:48 +0000  Wim Taymans <wim.taymans@gmail.com>

          tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
          Original commit message from CVS:
          * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
          Pushing 10 buffers is enough to run the test.

2008-12-11 10:28:43 +0000  Wim Taymans <wim.taymans@gmail.com>

          tests/examples/seek/seek.c: Hook up the SKIP seek flag.
          Original commit message from CVS:
          * tests/examples/seek/seek.c: (do_seek), (stop_cb),
          (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
          (main):
          Hook up the SKIP seek flag.

2008-12-10 18:43:32 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
          Original commit message from CVS:
          * gst/playback/gstplaybin2.c: (pad_added_cb):
          Error out with a missing-plugin error when the input-selector was not
          found.
          * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
          Indentation.

2008-12-10 17:39:32 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
          Original commit message from CVS:
          * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
          (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
          (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
          (gst_play_sink_send_event), (gst_play_sink_change_state):
          Use G_DEFINE_TYPE.
          Try to set the selected sink to READY before using it. This will allow
          for detection of incompatible formats sooner.
          Don't cause a fatal error when conversion elements are missing but post
          a missing-element message and a warning instead because things might
          still link and run fine.
          Simplyfy the construction of audio and video sink chains.

2008-12-10 14:55:10 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
          Original commit message from CVS:
          * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
          (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
          Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
          init from glib.

2008-12-10 08:19:13 +0000  Luis Menina <liberforce@freeside.fr>

          gst/: Include glib.h instead of a specific GLib header. Including single
          Original commit message from CVS:
          Patch by: Luis Menina <liberforce at freeside dot fr>
          * gst-libs/gst/floatcast/floatcast.h:
          * gst/typefind/gsttypefindfunctions.c:
          Include glib.h instead of a specific GLib header. Including single
          GLib headers is deprecated. Fixes bug #563904.

2008-12-09 18:30:10 +0000  Julien Moutte <julien@moutte.net>

          gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
          Original commit message from CVS:
          2008-12-09  Julien Moutte  <julien@fluendo.com>
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          Support higher max audio rates for some formats (WAV, Vorbis, LPCM).

2008-12-09 17:21:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-read.c:
          Fix handling of odd chunks in riff metadata.

2008-12-08 18:44:22 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
          Original commit message from CVS:
          * gst/volume/gstvolume.c: (gst_volume_class_init),
          (volume_before_transform), (volume_transform_ip):
          Use new basetransform vmethod to reconfigure the dynamic properties and
          any pending volume/mute changes. Fixes #563508.

2008-12-08 18:12:18 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
          Original commit message from CVS:
          * configure.ac:
          First check for "theoraenc theoradec" and if that failed check
          for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
          deprecate the latter. Also linking on Windows fails with just "theora"
          and the version check would fail for the release candidates.
          Fixes bug #563718.

2008-12-08 15:25:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
          Original commit message from CVS:
          * gst/playback/gstdecodebin.c:
          * gst/playback/gstdecodebin2.c:
          Add basic docs to decodebin and link to decodebin from decodebin2.

2008-12-08 12:08:32 +0000  Olivier Crete <tester@tester.ca>

          gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
          Original commit message from CVS:
          Patch by: Olivier Crete  <tester at tester ca>
          * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
          * gst-libs/gst/rtp/gstrtcpbuffer.h:
          Implement gst_rtcp_packet_remove(). Fixes #563174.
          * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
          Add unit test for some RTCP functions.

2008-12-04 20:09:19 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
          Original commit message from CVS:
          * configure.ac:
          Apparently AC_CONFIG_MACRO_DIR breaks when using more
          than one macro directory, reverting last change.

2008-12-04 19:47:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
          Original commit message from CVS:
          * configure.ac:
          Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
          our M4 macros.

2008-12-03 17:47:44 +0000  Edward Hervey <bilboed@bilboed.com>

          sys/: Clear all flags on buffers returned from the image pool.
          Original commit message from CVS:
          * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
          Clear all flags on buffers returned from the image pool.
          Fixes #563143

2008-12-01 19:36:35 +0000  이문형 <iwings@gmail.com>

          gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
          Original commit message from CVS:
          Patch by: 이문형 <iwings at gmail dot com>
          * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
          Don't forget to release the lock again if we bail out because some
          pad is flushing or we've reached EOS, otherwise things will lock up
          next time _push_buffer() is called (#562802).

2008-11-29 13:31:47 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
          Original commit message from CVS:
          Patch by: Cygwin Ports maintainer
          <yselkowitz at users dot sourceforge dot net>
          * autogen.sh:
          * configure.ac:
          Require gettext 0.17 because older versions don't mix with libtool
          2.2. At build time an older gettext version will still work.
          Fixes bug #556091.

2008-11-28 13:30:36 +0000  Christian Schaller <uraeus@gnome.org>

        * 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.

2008-11-24 20:25:24 +0000  Michael Smith <msmith@xiph.org>

          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.

2008-11-24 20:11:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-11-24 19:18:59 +0000  David Schleef <ds@schleef.org>

          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.

2008-11-24 12:56:54 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

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'.

2008-11-21 15:45:15 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-11-21 00:04:48 +0000  Michael Smith <msmith@xiph.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.

2008-11-20 22:11:38 +0000  Michael Smith <msmith@xiph.org>

          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.

2008-11-20 22:06:05 +0000  Michael Smith <msmith@xiph.org>

          gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
          Original commit message from CVS:
          * gst-libs/gst/rtsp/gstrtspdefs.c:
          Fix win32 build. Oops.

2008-11-20 21:40:49 +0000  Michael Smith <msmith@xiph.org>

          gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
          Original commit message from CVS:
          * gst-libs/gst/rtsp/gstrtspdefs.c:
          Use WSAGetLastError() rather than errno/h_errno on win32.

2008-11-20 21:20:27 +0000  Michael Smith <msmith@xiph.org>

          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.

2008-11-19 00:24:44 +0000  David Schleef <ds@schleef.org>

          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).

2008-11-18 18:08:42 +0000  Alessandro Decina <alessandro.d@gmail.com>

          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.

2008-11-14 21:44:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
          Original commit message from CVS:
          * gst/audioresample/gstaudioresample.c:
          Guard against a NULL dereference I somehow encountered -
          with a FLUSH_STOP arriving either before basetransform _start(),
          or after _stop().
          * gst/typefind/gsttypefindfunctions.c:
          Make sure we never jump backwards when typefinding corrupt mov files.

2008-11-14 21:39:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2008-11-14 15:40:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-11-13 21:11:13 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

          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.

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().

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.

2008-10-22 12:25:02 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
          Original commit message from CVS:
          * sys/xvimage/xvimagesink.c:
          * sys/xvimage/xvimagesink.h:
          * tests/icles/Makefile.am:
          * tests/icles/test-colorkey.c:
          Allow setting colorkey if possible. Implement property probe interface
          for optional X features (autopaint-colorkey, double-buffer and
          colorkey). Fixes #554533

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>

          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.

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.

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.

2008-10-08 15:30:33 +0000  Edward Hervey <bilboed@bilboed.com>

          gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosrc.c:
          (gst_base_audio_src_create):
          Fix debug statements (space between '%' and actual format).

2008-10-08 14:44:04 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-10-08 14:01:42 +0000  Andy Wingo <wingo@pobox.com>

        * ChangeLog:
          changelog
          Original commit message from CVS:
          changelog

2008-10-08 14:00:07 +0000  Andy Wingo <wingo@pobox.com>

          gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
          Original commit message from CVS:
          2008-10-08  Andy Wingo  <wingo@pobox.com>
          * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
          target instead of setting it.
          (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
          API for a decode pad. The bugfix is that we set the group in
          activate(), not when the pad was created because it might be NULL
          then.
          (gst_decode_group_control_source_pad, gst_decode_group_expose):
          Update to use the API.

2008-10-08 12:49:40 +0000  Andy Wingo <wingo@pobox.com>

          gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
          Original commit message from CVS:
          2008-10-08  Andy Wingo  <wingo@pobox.com>
          * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
          be a subclass of GstGhostPad.
          (analyze_new_pad): So, when emitting the signals that determine
          how we do autoplugging, already create the ghost pad and use it as
          the pad in the signal arguments. This allows applications to make
          a connection between the pad passed in e.g. autoplug-continue, and
          the pad passed in new-decoded-pad.
          (connect_pad, expose_pad): Update to receive the ghosted decode
          pad in the args, retargetting it as necessary if we have to plug
          the target pad through a multiqueue.
          (gst_decode_group_control_source_pad): Adapt to receive an
          already-ghosted pad that just needs activation, blocking, and
          drain notification.
          (sort_end_pads): Adapt for decode pads actually being pads.
          (gst_decode_group_expose): Adapt for decode pads actually being
          pads. Rewrite the decode pad names so they appear in order. Adds a
          new error case if we couldn't set the name.
          (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
          logic.
          (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
          New API for the decode pad, needed because we shouldn't do these
          things inside gst_decode_pad_new(), but after.
          (gst_decode_pad_new): Change to actually make the real pad, and
          delay the blocking/drainage bits.

2008-10-08 12:12:01 +0000  Daniel Drake <dsd@laptop.org>

          ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
          Original commit message from CVS:
          Patch by: Daniel Drake <dsd at laptop dot org>
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
          Unref all buffers when clearing collectpads. Fixes bug #546955.

2008-10-08 12:08:01 +0000  Klaas <klaas@rivercrew.net>

          ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
          Original commit message from CVS:
          Based on a patch by: Klaas <klaas at rivercrew dot net>
          * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
          (gst_vorbis_enc_buffer_check_discontinuous),
          (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
          * ext/vorbis/vorbisenc.h:
          Keep track of the upstream segments and use the running time on that
          segment instead of the buffer timestamp everywhere. Fixes bug #525807.

2008-10-08 11:50:50 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
          Original commit message from CVS:
          * gst/audioconvert/audioconvert.c: (audio_convert_convert):
          Prevent overflows with big buffer when calculating the size of
          the intermediate buffer by using gst_util_uint64_scale() instead of
          plain arithmetics. Fixes bug #552801.

2008-10-08 10:49:15 +0000  Pavel Zeldin <pzeldin@gmail.com>

          ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
          Original commit message from CVS:
          Patch by: Pavel Zeldin <pzeldin at gmail dot com>
          * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
          (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
          (gst_clock_overlay_init), (gst_clock_overlay_set_property),
          (gst_clock_overlay_get_property):
          * ext/pango/gstclockoverlay.h:
          API: Add ability to specify format for date/time display by
          adding a "time-format" property.
          Fixes bug #554879.

2008-10-08 09:22:26 +0000  Jan Gerber <j@oil21.org>

          gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
          Original commit message from CVS:
          Patch by: Jan Gerber <j at oil21 dot org>
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
          (gst_riff_create_video_template_caps):
          Add FFV1 fourcc to support playback of FFMPEG lossless video
          in AVI. Fixes bug #555319.

2008-10-08 09:12:36 +0000  Håvard Graff <havard.graff@tandberg.com>

          gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
          Original commit message from CVS:
          Patch by: Håvard Graff <havard dot graff at tandberg dot com>
          * gst-libs/gst/audio/gstbaseaudiosrc.c:
          (gst_base_audio_src_create):
          Implement skew clock slaving. Fixes #552559.

2008-10-08 09:10:23 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/: Fix include of config.h
          Original commit message from CVS:
          * gst-libs/gst/audio/multichannel.c:
          * gst-libs/gst/audio/testchannels.c:
          Fix include of config.h

2008-10-06 16:36:20 +0000  Tero Saarni <tero.saarni@gmail.com>

          gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
          Original commit message from CVS:
          Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
          * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
          (print_media), (gst_sdp_message_dump):
          Fix parsing of the c= field containing multicast addresses.
          Fixes #552199.
          Add the connection info to the session or streams.
          Fix parsing of the bandwidth.
          Add debugging for the connections and bandwidths for a media.
          Add debugging for the bandwidth of the session.

2008-10-06 16:31:27 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
          Original commit message from CVS:
          * gst-libs/gst/rtp/gstbasertppayload.c:
          (gst_basertppayload_change_state):
          Configure the next seqnum and timestamp in the state change so that they
          can be queried soon after.

2008-10-06 16:29:33 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
          Original commit message from CVS:
          * gst-libs/gst/rtp/gstbasertpdepayload.c:
          (gst_base_rtp_depayload_chain):
          Improve debugging of the rtptime.

2008-10-05 11:33:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Back to development -> 0.10.21.1
          Original commit message from CVS:
          * configure.ac:
          Back to development -> 0.10.21.1

2008-10-05 08:18:31 +0000  Sebastian Dröge <slomo@circular-chaos.org>

        * ChangeLog:
          ChangeLog surgery
          Original commit message from CVS:
          ChangeLog surgery

2008-10-05 08:11:53 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
          (plugin_init):
          Add typefinder for MXF.

2008-10-05 08:10:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
          (plugin_init):
          Add typefinder for MXF.

2008-10-03 15:19:40 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

=== release 0.10.21 ===

2008-10-03 00:03:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/gst-plugins-base-plugins.hierarchy:
        * docs/plugins/gst-plugins-base-plugins.interfaces:
        * docs/plugins/gst-plugins-base-plugins.prerequisites:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gdp.xml:
        * docs/plugins/inspect/plugin-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>

        * 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

2008-09-28 22:58:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: 0.10.20.4 pre-release
          Original commit message from CVS:
          * configure.ac:
          0.10.20.4 pre-release

2008-09-25 10:46:00 +0000  ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>

          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.

2008-09-23 17:48:14 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
          Original commit message from CVS:
          * gst-libs/gst/rtsp/gstrtspmessage.c:
          (gst_rtsp_message_parse_request),
          (gst_rtsp_message_parse_response):
          Fix the g_return_val_if_fail() statements.

2008-09-22 17:44:14 +0000  Michael Smith <msmith@xiph.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

2008-09-16 15:36:56 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          Commit stuff that should have gone in last week when I made the pre-releases:
          Original commit message from CVS:
          Commit stuff that should have gone in last week when I made the pre-releases:
          2008-09-10  Jan Schmidt  <jan.schmidt@sun.com>
          * configure.ac:
          0.10.20.2 pre-release
          * po/LINGUAS:
          * po/id.po:
          * po/pt_BR.po:
          New translations.

2008-09-15 15:11:18 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/: Recognise Kate subtitle streams (#550582).
          Original commit message from CVS:
          * gst-libs/gst/pbutils/descriptions.c:
          * gst/typefind/gsttypefindfunctions.c:
          Recognise Kate subtitle streams (#550582).

2008-09-13 11:04:02 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
          Original commit message from CVS:
          * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
          Remove trailing comma from enum list, which causes problems
          with -pendantic (#550729).

2008-09-05 19:04:47 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-09-05 14:12:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-09-05 10:24:05 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2008-09-04 16:25:06 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-09-04 13:46:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

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.

2008-09-02 15:07:09 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-09-02 09:33:17 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-09-02 06:37:04 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2008-09-01 16:05:45 +0000  Edward Hervey <bilboed@bilboed.com>

          tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
          Original commit message from CVS:
          * tests/check/elements/audioresample.c: (setup_audioresample),
          (fail_unless_perfect_stream), (test_perfect_stream_instance),
          (test_discont_stream_instance):
          Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
          Add debugging for coherence.

2008-08-30 15:55:06 +0000  Jonathan Matthew <notverysmart@gmail.com>

          gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
          Original commit message from CVS:
          Patch by: Jonathan Matthew  <notverysmart gmail com>
          * gst/typefind/gsttypefindfunctions.c: (plugin_init):
          Add typefinder for PDF documents (which is nice to have, since it's a
          common format, but also helps prevent false positives). Fixes #549814.

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.

2008-08-16 20:57:27 +0000  David Schleef <ds@schleef.org>

          gst/typefind/gsttypefindfunctions.c: DV typefinding.  Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
          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.

2008-08-15 07:24:38 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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".

2008-08-14 17:14:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-08-14 15:17:31 +0000  Julien Moutte <julien@moutte.net>

          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.

2008-08-13 09:17:38 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-08-13 00:59:07 +0000  David Schleef <ds@schleef.org>

          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.

2008-08-12 06:31:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
          Original commit message from CVS:
          * gst-libs/gst/interfaces/propertyprobe.h:
          Remove double "interface" from doc-string.
          * gst-libs/gst/interfaces/xoverlay.h:
          Document interface.
          * gst-libs/gst/riff/riff.c:
          Add basic doc blobs.

2008-08-11 15:05:35 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
          Original commit message from CVS:
          * gst-libs/gst/audio/Makefile.am:
          Don't try to build that example anymore.

2008-08-11 14:51:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
          Original commit message from CVS:
          * gst-libs/gst/audio/.cvsignore:
          * gst-libs/gst/audio/Makefile.am:
          * gst-libs/gst/audio/gstaudiofiltertemplate.c:
          * gst-libs/gst/audio/make_filter:
          Move audiofiltertemplate to gst-template.

2008-08-11 09:20:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          More docs and shuffling. What can we do with the hundreds of #defines.
          Original commit message from CVS:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          * gst-libs/gst/audio/gstaudiosrc.h:
          More docs and shuffling. What can we do with the hundreds of #defines.

2008-08-11 08:34:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-08-11 07:16:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-08-05 15:38:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          ext/vorbis/vorbisdec.c: Do not leak old taglist.
          Original commit message from CVS:
          * ext/vorbis/vorbisdec.c:
          Do not leak old taglist.

2008-08-04 12:35:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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().

2008-08-04 09:11:08 +0000  Andy Wingo <wingo@pobox.com>

          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.

2008-08-01 13:06:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-08-01 11:55:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-08-01 11:14:49 +0000  Wim Taymans <wim.taymans@gmail.com>

          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'.

2008-07-31 13:06:13 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-07-31 12:58:44 +0000  Wim Taymans <wim.taymans@gmail.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.

2008-07-31 11:39:44 +0000  Wim Taymans <wim.taymans@gmail.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.

2008-07-30 19:51:36 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
          Original commit message from CVS:
          * gst/audioconvert/Makefile.am:
          Dist recently-added gstfastrandom.h.

2008-07-30 15:29:44 +0000  Edward Hervey <bilboed@bilboed.com>

          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 (?).

2008-07-30 09:02:31 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-07-29 13:01:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
          Original commit message from CVS:
          * sys/xvimage/Makefile.am:
          floor() needs linking to $(LIBM).

2008-07-29 12:35:54 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2008-07-29 10:26:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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).

2008-07-29 08:59:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2008-07-29 01:58:05 +0000  David Schleef <ds@schleef.org>

          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.

2008-07-28 15:34:13 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-07-28 13:12:51 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-07-28 12:47:06 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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).

2008-07-27 11:12:41 +0000  Jan Gerber <j@oil21.org>

          ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
          Original commit message from CVS:
          Patch by: Jan Gerber <j at oil21 dot org>
          * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
          Fix calculation of the start time from skeleton streams.
          Fixes bug #530068.

2008-07-24 13:19:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

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.

2008-07-23 13:17:31 +0000  Damien Lespiau <damien.lespiau@gmail.com>

          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.

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.

2008-07-15 22:43:16 +0000  David Schleef <ds@schleef.org>

          gst-libs/gst/video/video.c: Revert ABI change.
          Original commit message from CVS:
          * gst-libs/gst/video/video.c: Revert ABI change.

2008-07-15 13:05:04 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-07-14 17:06:26 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

          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.

2008-07-14 08:18:58 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c:
          * gst/playback/gstplaybasebin.h:
          * gst/playback/gstplaybin.c:
          * gst/playback/gststreamselector.c:
          First stab at integrating DVD subpicture overlay into
          playbin. Successfully plugs and plays, but the queues need
          shrinking - 3 seconds of video is too much buffering.

2008-07-11 18:06:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

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.

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.

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.

2008-06-27 07:55:40 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          ChangeLog: ChangeLog surgery.
          Original commit message from CVS:
          * ChangeLog:
          ChangeLog surgery.
          * tests/examples/seek/seek.c:
          Move variable into ifdef too.

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().

2008-06-24 16:27:35 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2008-06-24 16:22:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
          Original commit message from CVS:
          * gst/playback/gstqueue2.c:
          Do not double notify. Remove the unsued return value.

2008-06-24 16:15:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
          Original commit message from CVS:
          * ext/alsa/gstalsamixer.c:
          Also consider "speaker" as a name for master volume. If that doesn't
          help look for the first non-mono volume control that also has a
          playback switch.

2008-06-24 16:10:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          ChangeLog: Forgot to save the ChangeLog :/
          Original commit message from CVS:
          * ChangeLog:
          Forgot to save the ChangeLog :/

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.

2008-06-21 18:56:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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.

2008-06-21 10:25:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
          break long lines
          Original commit message from CVS:
          break long lines

2008-06-20 18:24:24 +0000  Michael Smith <msmith@xiph.org>

          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.

2008-06-20 17:27:03 +0000  Michael Smith <msmith@xiph.org>

          gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
          Original commit message from CVS:
          * gst/playback/gstplaybin2.c:
          Use a different constant for the convert-frame signal id.
          Fixes #537009.

2008-06-20 17:18:55 +0000  Michael Smith <msmith@xiph.org>

          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.

2008-06-20 17:02:48 +0000  Michael Smith <msmith@xiph.org>

          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>

          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.

2008-06-20 16:12:50 +0000  Thomas Vander Stichele <thomas@apestaart.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.

=== release 0.10.20 ===

2008-06-18 14:36:28 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/gst-plugins-base-plugins.hierarchy:
        * docs/plugins/gst-plugins-base-plugins.interfaces:
        * docs/plugins/gst-plugins-base-plugins.prerequisites:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gdp.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playback.xml:
        * docs/plugins/inspect/plugin-queue2.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-uridecodebin.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * gst-plugins-base.doap:
        * po/LINGUAS:
        * win32/common/config.h:
          Release 0.10.20
          Original commit message from CVS:
          Release 0.10.20

2008-06-18 14:32:12 +0000  Jan Schmidt <thaytan@mad.scientist.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/it.po:
        * po/lt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/ru.po:
        * po/sk.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          Update .po files
          Original commit message from CVS:
          Update .po files

2008-06-18 06:31:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
          Original commit message from CVS:
          * docs/plugins/gst-plugins-bad-plugins-sections.txt:
          * examples/app/appsrc-ra.c:
          * examples/app/appsrc-seekable.c:
          * examples/app/appsrc-stream.c:
          * examples/app/appsrc-stream2.c:
          * ext/directfb/dfbvideosink.h:
          * ext/metadata/gstbasemetadata.c:
          * ext/metadata/gstbasemetadata.h:
          * ext/metadata/metadata.c:
          * ext/metadata/metadataexif.c:
          * ext/theora/theoradec.h:
          * gst/deinterlace2/gstdeinterlace2.h:
          * gst/deinterlace2/tvtime/speedy.c:
          * gst/deinterlace2/tvtime/speedy.h:
          * gst/deinterlace2/tvtime/vfir.c:
          Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
          comments.

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.

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.

2008-06-04 17:12:40 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
          (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
          (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
          (h264_video_type_find), (mpeg_video_stream_type_find),
          (dv_type_find), (mmsh_type_find):
          Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
          copy caps for no good reason (this may be desirable to make it easier
          to detect leaks, but then it should probably be done for all caps
          in the typefinder somewhere).

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.

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)

2008-06-03 19:44:48 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
          Original commit message from CVS:
          * gst-libs/gst/tag/tag.h:
          * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
          Don't forget to add new enum value here too (should probably use
          glib-mkenums here...).

2008-06-03 19:29:06 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
          Original commit message from CVS:
          * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
          * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
          * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
          (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
          (gst_tag_image_data_to_image_buffer):
          Add two utility functions to avoid code duplication (#512333):
          API: add gst_tag_image_data_to_image_buffer()
          API: add gst_tag_list_add_id3_image()

2008-06-03 08:54:29 +0000  Sebastian Dröge <slomo@circular-chaos.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.

2008-06-03 08:48:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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().

2008-06-02 20:09:14 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

2008-06-02 12:20:35 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-05-31 19:57:57 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

          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.

2008-05-31 19:50:59 +0000  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

        * 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.

2008-05-28 08:14:47 +0000  Sebastian Dröge <slomo@circular-chaos.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.

2008-05-27 18:10:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

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.

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 ';'.

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.

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.

2008-05-19 15:59:40 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-05-19 14:09:08 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

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.

2008-05-10 20:16:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-05-10 18:19:17 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

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.

2008-05-05 07:41:03 +0000  Edward Hervey <bilboed@bilboed.com>

          gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
          Original commit message from CVS:
          * gst-libs/gst/pbutils/descriptions.c: (formats):
          It's SorensOn and not SorensEn.

2008-05-04 15:23:36 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-05-04 15:02:20 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-05-03 16:00:04 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-05-03 15:45:23 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-05-03 15:39:04 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-05-03 12:09:16 +0000  Tim-Philipp Müller <tim@centricular.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).

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 :)

2008-05-02 10:54:51 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

        * 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.

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.

2008-04-30 17:06:45 +0000  Michael Smith <msmith@xiph.org>

          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.

2008-04-30 14:37:52 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
          (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
          (mpeg_video_stream_type_find):
          Refactor a bit: use context structure to track parsing offset and size of
          available data and make the code a bit clearer. Fixes bad memory access
          in #356937.

2008-04-28 22:18:49 +0000  Michael Smith <msmith@xiph.org>

          gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
          Original commit message from CVS:
          * gst/playback/test4.c:
          * gst/playback/test5.c:
          * gst/playback/test6.c:
          * gst/tcp/gstmultifdsink.c:
          Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
          is defined.

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

2008-04-23 13:50:34 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2008-04-23 13:45:29 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2008-04-23 08:58:42 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2008-04-22 06:18:04 +0000  David Schleef <ds@schleef.org>

          ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing.  Removes the weird "KW-DIRAC" bos packet.  Should conform to what we cu...
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c:
          Update Ogg/Dirac muxing.  Removes the weird "KW-DIRAC" bos
          packet.  Should conform to what we currently think is the
          final Ogg/Dirac muxing spec.

2008-04-22 06:13:43 +0000  David Schleef <ds@schleef.org>

          sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display.  Dark g...
          Original commit message from CVS:
          * sys/xvimage/xvimagesink.c:
          Fix typo that causes the overlay keying color to bright green
          on a 16-bit display.  Dark grey good.  Bright green bad.

2008-04-21 13:47:06 +0000  Stefan Kost <ensonic@users.sourceforge.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.

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.

2008-04-19 16:33:24 +0000  Sebastian Dröge <slomo@circular-chaos.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().

2008-04-18 17:10:43 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-04-18 14:54:01 +0000  Edward Hervey <bilboed@bilboed.com>

          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

2008-04-17 07:33:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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().

2008-04-17 07:29:28 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-04-15 19:06:00 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2008-04-15 19:02:10 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst/playback/gstdecodebin2.c: Fix signal docs.
          Original commit message from CVS:
          * gst/playback/gstdecodebin2.c:
          Fix signal docs.

2008-04-14 17:58:19 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-04-11 17:13:52 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
          Original commit message from CVS:
          * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
          (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
          (gst_video_format_get_pixel_stride),
          (gst_video_format_get_component_width),
          (gst_video_format_get_component_height),
          (gst_video_format_get_component_offset), (gst_video_format_get_size),
          (gst_video_format_convert):
          Add guards to these functions to ensure sane input values.
          * tests/check/libs/video.c:
          Fix unit test not to create caps with width=0 and height=0.

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>

        * 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.

2008-04-07 22:26:50 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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.

2008-04-07 10:50:11 +0000  Tim-Philipp Müller <tim@centricular.net>

          Merge other changes from 0.10.19 release branch.
          Original commit message from CVS:
          * NEWS:
          * RELEASE:
          * gst-plugins-base.doap:
          Merge other changes from 0.10.19 release branch.

2008-04-06 20:16:27 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_class_init):
          * gst-libs/gst/audio/gstbaseaudiosrc.c:
          (gst_base_audio_src_class_init):
          * gst/playback/gstplayback.c: (plugin_init):
          * gst/volume/gstvolume.c: (plugin_init):
          Work around missing bits of thread-safety on older GLibs some
          more to avoid assertions when starting up multiple playbin
          objects concurrently (see #512382).

2008-04-06 17:19:39 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-04-06 08:56:07 +0000  Damien Lespiau <damien.lespiau@gmail.com>

          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.

2008-04-03 23:01:11 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-04-03 16:10:53 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-04-03 15:58:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/ogg/gstoggmux.c: Fix sample pipeline description.
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c:
          Fix sample pipeline description.

2008-04-03 14:58:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

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.

2008-03-24 17:45:36 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/pango/gsttextoverlay.c: Fix Since: version for new property.
          Original commit message from CVS:
          * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
          Fix Since: version for new property.

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.

2008-03-24 11:24:22 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
          Original commit message from CVS:
          * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
          Small debug improvement.
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_render):
          Fix bug in determining the sample start/stop position, we want to base
          this decision on the fact that we are going forwards or backwards, not
          slower or faster. This fixes some ugly resync warnings when playing at
          very slow speeds.

2008-03-23 13:41:28 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-03-23 13:12:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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

=== release 0.10.18 ===

2008-03-21 00:26:03 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/gst-plugins-base-plugins.hierarchy:
        * docs/plugins/gst-plugins-base-plugins.interfaces:
        * docs/plugins/gst-plugins-base-plugins.prerequisites:
        * docs/plugins/gst-plugins-base-plugins.signals:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gdp.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playback.xml:
        * docs/plugins/inspect/plugin-queue2.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-uridecodebin.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * gst-plugins-base.doap:
        * po/LINGUAS:
        * win32/common/config.h:
          Release 0.10.18
          Original commit message from CVS:
          Release 0.10.18

2008-03-21 00:16:37 +0000  Jan Schmidt <thaytan@mad.scientist.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/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.

2008-03-17 10:32:28 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-03-14 18:42:35 +0000  David Schleef <ds@schleef.org>

          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.

2008-03-14 17:33:09 +0000  William M. Brack <wbrack@mmm.hk>

          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.

2008-03-14 09:54:44 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-03-12 12:39:13 +0000  Michael Smith <msmith@xiph.org>

          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.

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.

2008-03-03 13:56:38 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-03-03 13:50:18 +0000  Tim-Philipp Müller <tim@centricular.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)

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).

2008-03-02 18:32:36 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
          Original commit message from CVS:
          * gst-libs/gst/rtsp/gstrtspconnection.c:
          Don't include unix headers unconditionally (fixes #518037).

2008-03-02 18:24:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-03-02 18:20:44 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
          Original commit message from CVS:
          * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
          (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
          (gst_video_format_is_rgb), (gst_video_format_is_yuv),
          (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
          (gst_video_format_get_pixel_stride),
          (gst_video_format_get_component_width),
          (gst_video_format_get_component_height),
          (gst_video_format_get_component_offset), (gst_video_format_get_size):
          * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
          (GST_VIDEO_FORMAT_Y42B):
          API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.

2008-03-02 18:07:10 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2008-02-29 17:59:16 +0000  Tim-Philipp Müller <tim@centricular.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).

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>

        * 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().

2008-02-25 07:21:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-02-25 06:48:14 +0000  Sebastian Dröge <slomo@circular-chaos.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.

2008-02-25 06:29:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-02-23 09:51:26 +0000  Tomasz Sałaciński <tsalacinski@gmail.com>

          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).

2008-02-22 06:38:08 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-02-22 05:27:24 +0000  Brian Cameron <brian.cameron@sun.com>

          configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
          Original commit message from CVS:
          Patch by: Brian Cameron <brian dot cameron at sun dot com>
          * configure.ac:
          Don't hardcode -Wall and -Werror for configure checks, this fails
          with non-GCC compilers. Fixes bug #517991.

2008-02-21 08:05:10 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2008-02-20 15:37:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-02-20 11:52:28 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-02-19 20:42:09 +0000  Tim-Philipp Müller <tim@centricular.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'.

2008-02-19 20:36:58 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-02-19 20:22:09 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2008-02-19 16:16:55 +0000  Josep Torra Valles <josep@fluendo.com>

          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.

2008-02-19 15:50:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

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.

2008-02-18 13:51:34 +0000  Stefan Kost <ensonic@users.sourceforge.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.

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.

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)

2008-02-11 18:31:43 +0000  Tim-Philipp Müller <tim@centricular.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).

2008-02-11 18:02:13 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/gnomevfs/: Add support for https protocol. Fixes #510229.
          Original commit message from CVS:
          * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
          * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
          Add support for https protocol. Fixes #510229.

2008-02-11 17:03:18 +0000  Alan Peevers <peeves@pacbell.net>

          ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
          Original commit message from CVS:
          2008-02-11  Julien Moutte  <julien@fluendo.com>
          Patch by: Alan Peevers <peeves@pacbell.net>
          * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
          lock when calling alsa methods.

2008-02-11 13:03:13 +0000  Tim-Philipp Müller <tim@centricular.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).

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.

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.

2008-02-07 18:07:41 +0000  Christian Schaller <uraeus@gnome.org>

        * 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

2008-02-07 12:17:49 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
          Original commit message from CVS:
          * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
          Fix compiler warning.

2008-02-07 11:00:45 +0000  Peter Kjellerstedt <pkj@axis.com>

          gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
          Original commit message from CVS:
          Patch by: Peter Kjellerstedt  <pkj at axis com>
          * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
          Clear the addrinfo struct using memset. Fixes #514937.

2008-02-06 15:07:30 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/tcp/gstfdset.h: Remove unused field to same some memory.
          Original commit message from CVS:
          * gst/tcp/gstfdset.h:
          Remove unused field to same some memory.
          * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
          Mark action signals as such.

2008-02-06 13:35:58 +0000  Michael Smith <msmith@xiph.org>

          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.

2008-02-04 11:51:31 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2008-02-02 17:29:32 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2008-02-02 15:32:23 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

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.

2008-01-30 14:42:14 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Back to CVS
          Original commit message from CVS:
          * configure.ac:
          Back to CVS

=== release 0.10.17 ===

2008-01-30 14:19:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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.

2008-01-29 09:59:03 +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.

2008-01-29 09:47:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2008-01-28 23:35:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Back to CVS
          Original commit message from CVS:
          * configure.ac:
          Back to CVS

=== release 0.10.16 ===

2008-01-28 23:31:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/gst-plugins-base-plugins.hierarchy:
        * docs/plugins/gst-plugins-base-plugins.interfaces:
        * docs/plugins/gst-plugins-base-plugins.prerequisites:
        * docs/plugins/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

2008-01-28 22:15:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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.

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

2008-01-10 17:55:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_class_init):
          * gst-libs/gst/audio/gstbaseaudiosrc.c:
          (gst_base_audio_src_class_init):
          Ref audio clock class from a thread-safe context to make sure
          we're not bit by GObjects lack of thread-safety here (#349410),
          however unlikely that may be in practice.

2008-01-10 12:22:46 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
          Original commit message from CVS:
          * autogen.sh:
          Add -Wno-portability to the automake parameters to stop warnings
          about GNU make extensions being used. We require GNU make in almost
          every Makefile anyway.
          * configure.ac:
          Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
          at the same time is required for per target flags.

2008-01-08 21:10:02 +0000  Tim-Philipp Müller <tim@centricular.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).

2008-01-08 20:59:20 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-01-08 20:48:00 +0000  Tim-Philipp Müller <tim@centricular.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.

2008-01-07 13:59:43 +0000  Bastien Nocera <hadess@hadess.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).

2008-01-07 13:19:50 +0000  Jerone Young <jerone@gmail.com>

          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).

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.

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.

2007-12-31 21:31:01 +0000  David Schleef <ds@schleef.org>

          docs/: Add *-undeclared.txt to fix buildbot.
          Original commit message from CVS:
          * docs/libs/.cvsignore:
          * docs/plugins/.cvsignore:
          Add *-undeclared.txt to fix buildbot.

2007-12-31 20:45:28 +0000  David Schleef <ds@schleef.org>

          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.

2007-12-31 20:21:20 +0000  David Schleef <ds@schleef.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.

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).

2007-12-22 05:19:00 +0000  David Schleef <ds@schleef.org>

          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.

2007-12-21 22:46:56 +0000  Tim-Philipp Müller <tim@centricular.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).

2007-12-21 22:26:47 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-12-20 19:43:25 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

          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

2007-12-20 17:13:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/.cvsignore: Ignore more.
          Original commit message from CVS:
          * gst/playback/.cvsignore:
          Ignore more.

2007-12-20 10:41:29 +0000  Tim-Philipp Müller <tim@centricular.net>

          Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
          Original commit message from CVS:
          * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
          * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
          (set_active_source):
          * gst/playback/gstplaybasebin.h:
          * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
          (setup_sinks), (playbin_set_subtitles_visible):
          Make switching off of subtitles work. To avoid all kind of
          problems with unlinking of the subtitle input, we just keep
          the subtitle inputs linked as they are and tell textoverlay
          not to render them. Fixes #373011.
          Other subtitle switching issues (esp. when there are both
          external and in-stream subtitles) remain. They'll be solved
          in playbin2.

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.

2007-12-18 15:20:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2007-12-18 01:01:23 +0000  David Schleef <ds@schleef.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.

2007-12-18 00:13:26 +0000  David Schleef <ds@schleef.org>

          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.

2007-12-17 23:41:14 +0000  Tim-Philipp Müller <tim@centricular.net>

          Makefile.am: Use new common/win32.mak.
          Original commit message from CVS:
          * Makefile.am:
          Use new common/win32.mak.

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).

2007-12-16 23:46:16 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2007-12-15 17:27:48 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-12-15 03:40:34 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2007-12-14 19:06:24 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-12-14 18:46:12 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-12-14 16:23:06 +0000  Christian Schaller <uraeus@gnome.org>

        * 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

2007-12-14 09:24:55 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-12-13 15:54:00 +0000  Peter Kjellerstedt <pkj@axis.com>

          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.

2007-12-13 12:31:38 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-12-13 11:40:10 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
          Original commit message from CVS:
          * gst-libs/gst/pbutils/descriptions.c: (formats):
          Add description for 'private' dts caps (who come up with that name?).

2007-12-13 10:10:35 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-12-11 21:18:57 +0000  David Schleef <ds@schleef.org>

        * ChangeLog:
          ChangeLog: remove conflict markers
          Original commit message from CVS:
          ChangeLog: remove conflict markers

2007-12-11 17:14:13 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
          Original commit message from CVS:
          * ext/gnomevfs/Makefile.am:
          * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
          Use gst_tag_freeform_string_to_utf8() here, which also takes
          into account any character sets specified by the user via
          environment variables.

2007-12-10 15:21:41 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/audioconvert/Makefile.am: Also link to libm.
          Original commit message from CVS:
          * gst/audioconvert/Makefile.am:
          Also link to libm.

2007-12-10 15:13:55 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          No need for floating point operations here. avoids having to link
          against the math library too.

2007-12-10 11:16:25 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-12-09 04:28:38 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
          Original commit message from CVS:
          * configure.ac:
          Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.

2007-12-08 18:38:39 +0000  Robin Stocker <robin.stocker@gmx.ch>

          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.

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.

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.

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).

2007-11-16 01:21:40 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst-plugins-base.doap: Oops, fix the release name.
          Original commit message from CVS:
          * gst-plugins-base.doap:
          Oops, fix the release name.

2007-11-16 00:44:58 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst-plugins-base.doap: Add 0.10.15 release
          Original commit message from CVS:
          * gst-plugins-base.doap:
          Add 0.10.15 release

2007-11-16 00:24:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Back to CVS
          Original commit message from CVS:
          * configure.ac:
          Back to CVS

=== release 0.10.15 ===

2007-11-16 00:14:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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"

2007-11-16 00:04:24 +0000  Jan Schmidt <thaytan@mad.scientist.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/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.

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).

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).

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).

2007-11-03 10:39:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
          Original commit message from CVS:
          * ext/alsa/gstalsasink.c: (gst_alsasink_open):
          * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
          'Could not open resource for writing' is not an acceptable
          error message when we can't open the audio device (see #492334),
          even less so when we're trying to open it to record something.

2007-11-02 21:03:01 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

          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).

2007-11-02 14:59:06 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

          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.

2007-11-01 13:28:59 +0000  Vincent Torri <vtorri@univ-evry.fr>

          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).

2007-11-01 12:51:57 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

          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).

2007-11-01 08:06:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-10-31 17:54:48 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
          Original commit message from CVS:
          * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
          (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
          Preserve channel layout when fixating the number of channels in the
          output caps, or make sure there's a suitable channel position layout
          set on the caps if required. Fixes #430677.

2007-10-31 17:32:22 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-10-31 15:30:15 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-10-31 12:47:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-10-31 12:30:28 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-10-30 20:32:14 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2007-10-30 19:46:02 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/icles/stress-playbin.c: Fix the build.
          Original commit message from CVS:
          * tests/icles/stress-playbin.c:
          Fix the build.

2007-10-30 15:54:46 +0000  Tim-Philipp Müller <tim@centricular.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.

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>

        * 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.

2007-10-26 12:07:14 +0000  Christian Schaller <uraeus@gnome.org>

        * 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).

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).

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.

2007-10-15 11:38:39 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-10-12 10:52:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-10-12 10:37:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
          Remove stray character from the changelog.
          Original commit message from CVS:
          Remove stray character from the changelog.

2007-10-12 10:33:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
          I'm too lazy to comment this
          Original commit message from CVS:
          *** empty log message ***

2007-10-11 18:24:09 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-10-11 16:12:21 +0000  Jason Kivlighn <jkivlighn@gmail.com>

          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).

2007-10-10 17:01:51 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

2007-10-06 15:04:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
          Original commit message from CVS:
          * gst-libs/gst/tag/Makefile.am:
          * gst-libs/gst/tag/gsttagdemux.c:
          * gst-libs/gst/tag/gsttagdemux.h:
          API: add GstTagDemux base class for simple tag demuxers.
          * docs/libs/gst-plugins-base-libs-docs.sgml:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          Add GstTagDemux to docs.

2007-10-05 07:49:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
          Original commit message from CVS:
          * gst-libs/gst/rtp/gstrtpbuffer.c:
          (gst_rtp_buffer_get_payload_subbuffer):
          Fix bug introduced with last commit which inverted the logic and
          caused all buffers to be dropped. Fixes #483620.
          Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.

2007-10-04 06:50:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-10-03 14:51:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
          Original commit message from CVS:
          * tests/check/pipelines/simple-launch-lines.c:
          Print message name and not just number.

2007-10-02 11:11:13 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_async_play):
          When slaved to the clock, don't try to align a sample with the previous
          one when going to PLAYING again.

2007-10-02 09:04:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/examples/snapshot/snapshot.c: Fix the build.
          Original commit message from CVS:
          * tests/examples/snapshot/snapshot.c:
          Fix the build.

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.

2007-09-25 11:54:09 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-09-25 07:50:59 +0000  Sebastian Dröge <slomo@circular-chaos.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.

2007-09-22 17:58:22 +0000  Thomas Vander Stichele <thomas@apestaart.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

2007-09-22 17:57:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * po/LINGUAS:
          add languages
          Original commit message from CVS:
          add languages

2007-09-22 17:56:28 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          po/pl.po: Added Polish translation.
          Original commit message from CVS:
          translated by: Jakub Bogusz <qboosh@pld-linux.org>
          * po/pl.po:
          Added Polish translation.

2007-09-22 17:55:37 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          po/fi.po: Added Finnish translation.
          Original commit message from CVS:
          translated by: Ilkka Tuohela <hile@iki.fi>
          * po/fi.po:
          Added Finnish translation.

2007-09-22 17:54:50 +0000  Thomas Vander Stichele <thomas@apestaart.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.

2007-09-22 17:53:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          po/da.po: Added Danish translation.
          Original commit message from CVS:
          translated by: Mogens Jaeger <mogens@jaeger.tf>
          * po/da.po:
          Added Danish translation.

2007-09-22 17:52:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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.

2007-09-22 17:51:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          po/bg.po: Added Bulgarian translation.
          Original commit message from CVS:
          translated by: Alexander Shopov <ash@contact.bg>
          * po/bg.po:
          Added Bulgarian translation.

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.

2007-09-19 18:16:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          Ignore registries in any format.
          Original commit message from CVS:
          * docs/plugins/.cvsignore:
          * tests/check/.cvsignore:
          Ignore registries in any format.

2007-09-19 16:09:57 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-09-19 15:55:08 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-09-19 12:04:21 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2007-09-17 17:24:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

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.

2007-09-12 17:15:12 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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).

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.

2007-09-09 10:25:43 +0000  René Stadler <mail@renestadler.de>

          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.

2007-09-09 04:08:48 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2007-09-07 17:37:03 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-09-07 16:46:05 +0000  Haakon Sporsheim <haakon.sporsheim@tandberg.com>

          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).

2007-09-07 15:05:24 +0000  Christian Schaller <uraeus@gnome.org>

        * 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.

2007-09-06 20:31:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          configure.ac: Fix configure check for HAVE_LIBXML_HTML.
          Original commit message from CVS:
          * configure.ac:
          Fix configure check for HAVE_LIBXML_HTML.

2007-09-06 12:14:25 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-09-05 14:01:25 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gststreaminfo.c: Fix build.
          Original commit message from CVS:
          * gst/playback/gststreaminfo.c:
          Fix build.

2007-09-05 10:32:09 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
          Original commit message from CVS:
          * gst/playback/gststreaminfo.c:
          Clean up some half-disabled code and comment.

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.

2007-08-30 06:58:46 +0000  Daniel Díaz <yosoy@danieldiaz.org>

          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.

2007-08-29 14:22:04 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-08-29 12:16:46 +0000  Tim-Philipp Müller <tim@centricular.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).

2007-08-28 14:58:17 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-08-28 14:23:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-08-27 11:59:56 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-08-26 14:14:33 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

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

2007-08-23 08:33:43 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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:

2007-08-22 15:29:04 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-08-22 11:20:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

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.

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.

2007-08-13 15:26:54 +0000  Dan Williams <dcbw@redhat.com>

          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.

2007-08-13 11:21:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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)

2007-08-10 17:35:52 +0000  Stefan Kost <ensonic@users.sourceforge.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).

2007-08-10 13:55:44 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-08-10 10:08:05 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/: Printf format fixes (#465028).
          Original commit message from CVS:
          * gst/playback/gstqueue2.c:
          * gst/videorate/gstvideorate.c:
          Printf format fixes (#465028).

2007-08-09 15:44:02 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-08-09 12:06:43 +0000  Josep Torra Valles <josep@fluendo.com>

          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.

2007-08-09 11:37:28 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2007-08-09 10:51:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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.

2007-08-08 16:07:21 +0000  Yang Hong <hongyang@redflag-linux.com>

          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

2007-08-08 15:05:22 +0000  Josep Torre Valles <josep@fluendo.com>

          Add connection-speed property. Fixes #464690.
          Original commit message from CVS:
          Patch by: Josep Torre Valles <josep@fluendo.com>
          * docs/plugins/gst-plugins-base-plugins.args:
          * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
          (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
          (gst_uri_decode_bin_get_property), (gen_source_element):
          Add connection-speed property. Fixes #464690.

2007-08-07 15:13:46 +0000  Damien Lespiau <damien.lespiau@gmail.com>

          Fix compilation on windows. Fixes #464320.
          Original commit message from CVS:
          Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
          * configure.ac:
          * gst-libs/gst/rtsp/Makefile.am:
          * gst-libs/gst/rtsp/gstrtspconnection.c:
          (gst_rtsp_connection_connect):
          Fix compilation on windows. Fixes #464320.

2007-08-07 14:14:54 +0000  Josep Torre Valles <josep@fluendo.com>

          gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
          Original commit message from CVS:
          Patch by: Josep Torre Valles <josep@fluendo.com>
          * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
          (gst_play_base_bin_init), (queue_threshold_reached),
          (gen_source_element), (setup_substreams),
          (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
          (gst_play_base_bin_get_streaminfo_value_array):
          * gst/playback/gstplaybasebin.h:
          * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
          (gst_play_bin_set_property), (gst_play_bin_get_property),
          (gst_play_bin_handle_redirect_message):
          Move connection-speed property from playbin to playbasebin so that we
          can also configure it in source elements that have the connection-speed
          property. Fixes #464028.
          Add some debug info here and there.

2007-08-06 16:42:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
          Original commit message from CVS:
          * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
          Properly respond to conversion queries. Fixes #464079.

2007-08-03 19:53:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
          Original commit message from CVS:
          * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
          (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
          (gst_audio_test_src_init_sine_table),
          (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
          * gst/audiotestsrc/gstaudiotestsrc.h:
          Add float/double and int32 support to audiotestsrc. Fixes #460422.
          Also set the default volume to the default value specified in the
          GParamSpec.

2007-08-03 19:40:14 +0000  Jens Granseuer <jensgr@gmx.net>

          gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
          Original commit message from CVS:
          Patch by: Jens Granseuer <jensgr at gmx dot net>
          * gst/audioconvert/gstaudioquantize.c:
          Fix C89 incompatibilities and spelling of explanations. Fixes #463215.

2007-08-03 15:44:01 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
          Original commit message from CVS:
          * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
          Add rdt manager for rdt transport.
          Fix parsing of RDT transport.

2007-08-03 14:43:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Back to CVS
          Original commit message from CVS:
          * configure.ac:
          Back to CVS

=== release 0.10.14 ===

2007-08-03 14:41:46 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-decodebin2.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gdp.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playbin.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * gst-plugins-base.doap:
        * win32/common/config.h:
          Release 0.10.14
          Original commit message from CVS:
          Release 0.10.14

2007-08-03 14:24:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * po/af.po:
        * po/az.po:
        * po/cs.po:
        * po/de.po:
        * po/en_GB.po:
        * po/hu.po:
        * po/it.po:
        * po/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.

2007-07-23 11:18:35 +0000  Dan Williams <dcbw@redhat.com>

          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.

2007-07-23 10:41:18 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-07-20 07:22:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/: We should use $(LIBM).
          Original commit message from CVS:
          * tests/examples/volume/Makefile.am:
          * tests/icles/Makefile.am:
          We should use $(LIBM).

2007-07-20 06:13:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/icles/Makefile.am: This needs -lm.
          Original commit message from CVS:
          * tests/icles/Makefile.am:
          This needs -lm.

2007-07-18 07:35:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          Add stdlib include (free, atoi, exit).
          Original commit message from CVS:
          * examples/app/appsrc_ex.c:
          * examples/switch/switcher.c:
          * ext/neon/gstneonhttpsrc.c:
          * ext/timidity/gstwildmidi.c:
          * ext/x264/gstx264enc.c:
          * gst/mve/mveaudioenc.c: (mve_compress_audio):
          * gst/rtpmanager/gstrtpclient.c:
          * gst/rtpmanager/gstrtpjitterbuffer.c:
          * gst/spectrum/demo-audiotest.c:
          * gst/spectrum/demo-osssrc.c:
          * sys/dvb/gstdvbsrc.c:
          Add stdlib include (free, atoi, exit).

2007-07-16 10:10:28 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
          Original commit message from CVS:
          * gst-libs/gst/rtp/gstbasertppayload.c:
          (gst_basertppayload_class_init), (gst_basertppayload_init),
          (gst_basertppayload_set_property),
          (gst_basertppayload_get_property):
          Don't break ABI, restore previous ranges. Keep the default random
          selection of timestamp and seqnum offset but as soon as the app sets a
          specific value, use that one.

2007-07-14 18:33:15 +0000  Bastien Nocera <hadess@hadess.net>

          sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
          Original commit message from CVS:
          Patch by: Bastien Nocera <hadess at hadess dot net>
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
          (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
          (gst_xvimagesink_init), (gst_xvimagesink_class_init):
          * sys/xvimage/xvimagesink.h:
          Add option to turn off double-buffering for debugging purposes.
          Fixes #437169.

2007-07-14 18:20:41 +0000  Jorn Baayen <jorn@openedhand.com>

          sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
          Original commit message from CVS:
          Patch by: Jorn Baayen <jorn at openedhand dot com>
          * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
          (gst_ximagesink_set_property), (gst_ximagesink_get_property),
          (gst_ximagesink_init), (gst_ximagesink_class_init):
          * sys/ximage/ximagesink.h:
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
          (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
          (gst_xvimagesink_init), (gst_xvimagesink_class_init):
          * sys/xvimage/xvimagesink.h:
          add 'handle-expose' property. Useful for video widgets which may want to
          be in control of Expose behaviour. Fixes #380625

2007-07-14 17:23:42 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
          Original commit message from CVS:
          * gst-libs/gst/rtp/gstbasertppayload.c:
          (gst_basertppayload_class_init), (gst_basertppayload_init),
          (gst_basertppayload_event), (gst_basertppayload_push),
          (gst_basertppayload_set_property),
          (gst_basertppayload_get_property),
          (gst_basertppayload_change_state):
          * gst-libs/gst/rtp/gstbasertppayload.h:
          Fix ranges of rtp payloader properties so that the full range can be
          used in addition to -1 (random).
          Fix wrong seqnum reporting in caps.
          Fixes #420326.

2007-07-13 18:12:19 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/videorate/gstvideorate.c: Use boilerplate.
          Original commit message from CVS:
          * gst/videorate/gstvideorate.c: (gst_video_rate_init),
          (gst_video_rate_query):
          Use boilerplate.
          Add latency query, might not be perfect yet but already works a lot
          better. Fixes #442557.

2007-07-13 16:05:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
          Original commit message from CVS:
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
          (gst_xvimagesink_setcaps):
          * sys/xvimage/xvimagesink.h:
          After a caps change, redraw our borders to avoid garbage left there
          when the image format changes to a smaller size, like 16:9 -> 4:3
          Also, hold the flow_lock a bit longer in the set_caps while we're
          fiddling with the xcontext.

2007-07-13 16:02:23 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
          Original commit message from CVS:
          * Makefile.am:
          * configure.ac:
          * tests/Makefile.am:
          Remove bogus check for libcheck, since we check for
          gstreamer-check and it pulls in the required info from there, and we
          weren't actually _using_ the information for libcheck ourselves
          anyway.

2007-07-13 15:52:02 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
          Original commit message from CVS:
          * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
          (gst_ffmpeg_caps_to_pixfmt):
          Fix the r_mask test for RGBA32 on little-endian.
          Fix a stupid typo that would have obviously broken
          compilation on big-endian, if anyone was testing.

2007-07-12 15:02:43 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
          Original commit message from CVS:
          * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
          (paint_hline_str4):
          * gst/videotestsrc/videotestsrc.h:
          Add alpha to the color struct.
          Use a default alpha value of 255 instead of 128.

2007-07-12 12:01:20 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (no_more_pads_full),
          (setup_source):
          Clear the dynamic pads counter when starting a new uri. This makes
          reusing playbin work again.
          Fixes #454264.

2007-07-12 11:13:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          configure.ac: Use pkg-config to locate check.
          Original commit message from CVS:
          * configure.ac:
          Use pkg-config to locate check.

2007-07-11 23:12:12 +0000  Tim-Philipp Müller <tim@centricular.net>

          Fix 'make check' build against core CVS.
          Original commit message from CVS:
          * configure.ac:
          * tests/check/elements/volume.c: (GST_START_TEST):
          Fix 'make check' build against core CVS.

2007-07-10 20:46:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/: Make gtk-doc happy.
          Original commit message from CVS:
          * gst-libs/gst/interfaces/propertyprobe.c:
          * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
          * gst-libs/gst/tag/gstvorbistag.c:
          Make gtk-doc happy.

2007-07-08 13:07:38 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-07-06 18:19:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
          Original commit message from CVS:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          Fix location of includes in the docs.

2007-07-06 11:40:45 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
          Original commit message from CVS:
          * gst/ffmpegcolorspace/avcodec.h:
          * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
          (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
          (gst_ffmpegcsp_avpicture_fill):
          * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
          (img_get_alpha_info):
          Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
          of the existing BGRA32 and RGBA32 formats with the alpha at the other
          end of the word. Partially fixes #451908

2007-07-05 08:43:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          docs/: Simplify --extra-dir as gtkdoc scans recursively.
          Original commit message from CVS:
          * docs/libs/Makefile.am:
          * docs/plugins/Makefile.am:
          Simplify --extra-dir as gtkdoc scans recursively.

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.

2007-06-28 11:06:56 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-06-28 10:21:19 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-06-27 12:55:20 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
          Original commit message from CVS:
          * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
          Add description for Windows Media RTP caps.
          * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
          Remove RTP fields that don't define the format from caps.

2007-06-27 10:14:03 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-06-25 12:04:15 +0000  Tim-Philipp Müller <tim@centricular.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).

2007-06-23 14:44:07 +0000  Edward Hervey <bilboed@bilboed.com>

          ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
          Original commit message from CVS:
          * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
          The chain should be freed if we error out here, else it will leak.
          * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
          (cleanup_decodebin):
          Don't forget to *properly* remove the signals, else it will leak.

2007-06-22 14:25:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          MAINTAINERS: Updating all the maintainers files
          Original commit message from CVS:
          * MAINTAINERS:
          Updating all the maintainers files

2007-06-21 08:34:46 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
          Original commit message from CVS:
          * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
          (main):
          Destroy and recreate parse-launch based pipeline after stop to be able
          to play again. Reorder some code and add more comments.

2007-06-20 11:09:03 +0000  Wim Taymans <wim@fluendo.com>

          gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
          Original commit message from CVS:
          * gst/playback/gstdecodebin2.c: (analyze_new_pad):
          When handling a delayed-caps notification case, mark
          the group as dynamic so that the nbdynamic count is
          incremented and decremented correctly. Fixes: #449156
          Patch by: Wim Taymans <wim@fluendo.com>

2007-06-19 19:13:04 +0000  Andy Wingo <wingo@pobox.com>

        * 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.

2007-06-19 09:34:35 +0000  Michael Smith <msmith@xiph.org>

          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.

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.

2007-06-16 03:42:14 +0000  David Schleef <ds@schleef.org>

          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.

2007-06-15 15:23:36 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-06-15 11:15:28 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-06-14 19:53:27 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

2007-06-07 14:25:32 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
          The BOS page of the first Dirac video stream needs to come before
          the BOS page of any Vorbis streams or other audio streams, just like
          it is with Theora.

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.

2007-06-05 17:08:04 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-06-05 17:02:13 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-06-05 16:37:09 +0000  Edward Hervey <edward@fluendo.com>

          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.

2007-06-05 16:20:44 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
          Original commit message from CVS:
          * gst-libs/gst/riff/Makefile.am:
          * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
          Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
          our own implementation.

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 0.10.13 ===

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/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
          Update .po files
          Original commit message from CVS:
          Update .po files

2007-05-31 17:08:58 +0000  Wim Taymans <wim@fluendo.com>

          gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          In riff, the depth is stored in the size field but it just means that
          the least significant bits are cleared. We can therefore just play
          the sample as if it had a depth == width. Fixes: #440997
          Patch by: Wim Taymans <wim@fluendo.com>
          Patch by: Sebastian Dröge  <slomo@circular-chaos.org>

2007-05-31 16:36:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
          Original commit message from CVS:
          * gst-libs/gst/floatcast/floatcast.h:
          Define inline when needed on win32 builds. Fixes: #441295

2007-05-29 13:38:35 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (queue_overrun),
          (no_more_pads_full):
          Stop buffering when the group is commited because the queues filled up.
          Fixes #442024.

2007-05-25 10:07:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
          Original commit message from CVS:
          * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
          (gst_alsa_mixer_free), (gst_alsa_mixer_update),
          (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
          (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
          (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
          * ext/alsa/gstalsamixer.h:
          * ext/alsa/gstalsamixerelement.c:
          (gst_alsa_mixer_element_interface_supported),
          (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
          (gst_alsa_mixer_element_set_property),
          (gst_alsa_mixer_element_get_property),
          (gst_alsa_mixer_element_change_state):
          * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
          * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
          (gst_mixer_option_changed):
          * gst-libs/gst/interfaces/mixer.h:
          Revert commits towards #152864 made so far. We'll pick it up again
          after the 0.10.13 release.

2007-05-24 16:22:23 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_render):
          After an interrupt (PAUSED/flush) assume that the next sample should not
          be aligned to the previous sample. Fixes #417992.

2007-05-24 15:16:59 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          Don't add channels and rate fields to the template caps for
          audio/x-dts, as wavparse might not always be able to set them,
          which would then lead to 'caps are not a real subset of the
          template caps' warnings.

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.

2007-05-22 15:30:26 +0000  Mark Nauwelaerts <manauw@skynet.be>

          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.

2007-05-22 15:04:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
          Original commit message from CVS:
          * ext/cdparanoia/gstcdparanoiasrc.c:
          Repair umlaut.

2007-05-22 11:40:31 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          Specify the full valid range for MP3 samplerates. Fixes a regression
          caused by extra header checks since the last release.

2007-05-21 15:32:42 +0000  Mike Smith <msmith@xiph.org>

          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.

2007-05-21 15:24:21 +0000  Michael Smith <msmith@xiph.org>

          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)

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-plugins-base.spec.in:
          update spec
          Original commit message from CVS:
          update spec

2007-05-18 15:23:43 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/audio/gstbaseaudiosink.c
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c
          (gst_base_audio_sink_change_state):
          Fix typo in comment.
          * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
          free_dynamics, pad_probe, close_pad_link, try_to_link_1,
          get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
          close_link):
          * gst/playback/gstplaybin.c (gst_play_bin_set_property,
          gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
          Remove trailing whitespaces in comments.
          * gst/volume/Makefile.am:
          Fix tabs.

2007-05-18 15:10:08 +0000  Marc-Andre Lureau <marcandre.lureau@gmail.com>

        * 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).

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.

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.

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.

2007-05-15 20:14:06 +0000  David Schleef <ds@schleef.org>

          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.

2007-05-15 17:11:09 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-05-15 16:46:10 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/ogg/gstoggdemux.c: Add some FIXMEs.
          Original commit message from CVS:
          * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
          (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
          (gst_ogg_demux_handle_page):
          Add some FIXMEs.
          Fix chain start/stop segment handling based on patch by
          <ahalda at cs dot mcgill dot ca> see #320984.

2007-05-15 15:33:54 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-05-15 15:29:17 +0000  Stefan Kost <ensonic@users.sourceforge.net>

        * 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.

2007-05-15 14:01:26 +0000  Marc-Andre Lureau <marcandre.lureau@gmail.com>

          ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
          Original commit message from CVS:
          patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
          * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
          _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
          gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
          gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
          gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
          gst_alsa_mixer_handle_source_callback,
          gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
          gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
          gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
          gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
          gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
          gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
          * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
          * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
          gst_alsa_mixer_element_interface_supported,
          gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
          gst_alsa_mixer_element_set_property,
          gst_alsa_mixer_element_get_property,
          gst_alsa_mixer_element_change_state):
          * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
          * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
          gst_mixer_option_changed):
          * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
          volume_changed, option_changed, _gst_reserved):
          Implement notification for alsamixer. Fixes #152864

2007-05-15 03:53:11 +0000  David Schleef <ds@schleef.org>

          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.

2007-05-13 01:06:19 +0000  David Schleef <ds@schleef.org>

          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.

2007-05-12 16:18:39 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-05-12 16:16:22 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-05-11 17:33:43 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-05-09 11:54:32 +0000  Michael Smith <msmith@xiph.org>

          ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
          Original commit message from CVS:
          * ext/theora/theoradec.c: (theora_handle_type_packet):
          gst_pad_push_event() does not return a GstFlowReturn!

2007-05-09 11:25:34 +0000  Wim Taymans <wim.taymans@gmail.com>

          tests/examples/seek/: Some small cosmetic changes.
          Original commit message from CVS:
          * tests/examples/seek/scrubby.c: (stop_cb), (main):
          * tests/examples/seek/seek.c: (do_seek):
          Some small cosmetic changes.

2007-05-08 19:24:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

        * ChangeLog:
        * gst/adder/gstadder.c:
        * gst/adder/gstadder.h:
          gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
          Original commit message from CVS:
          * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
          gst_adder_change_state):
          * gst/adder/gstadder.h (bps, offset, collect_event, segment,
          segment_pending, segment_position, segment_rate):
          Handle playback-rate on adder.

2007-05-07 11:43:31 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-05-04 13:10:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-05-04 12:41:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-05-04 09:06:38 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-05-04 08:46:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-05-03 16:29:10 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-05-03 13:16:21 +0000  Michael Smith <msmith@xiph.org>

          gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
          Original commit message from CVS:
          * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
          If a buffer doesn't have a timestamp, assume it's contiguous with
          the previous buffer, and synthesise timestamps appropriately.

2007-05-03 11:24:00 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2007-05-03 10:47:22 +0000  Edward Hervey <bilboed@bilboed.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.

2007-05-01 18:45:36 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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).

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).

2007-04-24 18:58:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2007-04-24 15:00:07 +0000  Dan Williams <dcbw@redhat.com>

          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.

2007-04-23 20:04:28 +0000  Stefan Kost <ensonic@users.sourceforge.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

2007-04-21 15:29:27 +0000  Tim-Philipp Müller <tim@centricular.net>

        * 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).

2007-04-21 15:10:25 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-04-21 14:40:45 +0000  Zeeshan Ali <zeenix@gmail.com>

          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).

2007-04-21 14:14:24 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
          Original commit message from CVS:
          * gst/audioresample/gstaudioresample.c:
          Make more functions static, just because we can.

2007-04-21 13:54:39 +0000  Tim-Philipp Müller <tim@centricular.net>

          tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
          Original commit message from CVS:
          * tests/check/elements/audioresample.c:
          Add unit test for audioresample shutdown crasher (#420106).

2007-04-20 10:42:24 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst/subparse/: Use GST_DISABLE_XML here
          Original commit message from CVS:
          * gst/subparse/gstsubparse.c:
          * gst/subparse/samiparse.c:
          Use GST_DISABLE_XML here
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
          (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
          (gst_xvimagesink_buffer_alloc),
          (gst_xvimagesink_navigation_send_event):
          * sys/xvimage/xvimagesink.h:
          Include stdlib.h when using atoi.
          * tests/check/elements/playbin.c: (playbin_suite):
          Use GST_DISABLE_REGISTRY here

2007-04-19 16:58:53 +0000  Michael Smith <msmith@xiph.org>

          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).

2007-04-18 11:06:42 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/check/pipelines/.cvsignore: Fix build.
          Original commit message from CVS:
          * tests/check/pipelines/.cvsignore:
          Fix build.

2007-04-17 10:56:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
          Original commit message from CVS:
          * gst/app/Makefile.am:
          Fix CFLAGS and hopefully #430594.

2007-04-17 02:53:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          Allow random depths between 1 and 32 instead of only multiplies of 8.

2007-04-17 02:04:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          Set the maximum number of channels for PCM and float in the correct
          place to have it also used when creating the template caps.

2007-04-17 01:56:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          Correctly support 4, 6 and 8 channels with normal PCM and float
          wav files.
          Fix the depth and signedness calculation in extensible wav files and
          also handle 1, 2, 4, 6, 8 channels here when a file without channel
          mask is found.
          Add support for float, alaw and mulaw in extensible wav files.
          This allows correct playback of all but 5 files from
          http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
          (gst_riff_create_audio_template_caps):
          Add voxware and float formats to the template caps.

2007-04-16 22:20:03 +0000  Vincent Torri <vtorri@univ-evry.fr>

          ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
          Original commit message from CVS:
          Patch by: Vincent Torri <vtorri at univ-evry dot fr>
          * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
          Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
          Use the correct format strings for integer formats.

2007-04-16 21:44:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst-plugins-base.doap:
          fix release date
          Original commit message from CVS:
          fix release date

2007-04-16 21:42:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst-plugins-base.doap:
          fix release date
          Original commit message from CVS:
          fix release date

2007-04-15 14:35:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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.

2007-04-14 12:34:55 +0000  Thomas Vander Stichele <thomas@apestaart.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.

2007-04-13 22:10:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/tcp/gstmultifdsink.c:
          add debug
          Original commit message from CVS:
          add debug

2007-04-13 21:55:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * tests/check/pipelines/streamheader.c:
          tests/check/pipelines/streamheader.c (tag_event_probe_cb,
          Original commit message from CVS:
          * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
          GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
          streamheader_suite):
          Add another test set up for failure

2007-04-13 21:09:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ext/ogg/gstoggmux.c:
        * gst/gdp/gstgdpdepay.c:
          debug changes
          Original commit message from CVS:
          debug changes

2007-04-13 21:08:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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.

2007-04-13 11:42:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          ext/theora/theoradec.c: Fix misleading comment.
          Original commit message from CVS:
          * ext/theora/theoradec.c: (theora_dec_sink_event):
          Fix misleading comment.

2007-04-13 06:17:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-04-12 16:36:36 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
          Original commit message from CVS:
          * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
          Try encodings from all environment variables, not just those in the
          first environment variable that is set.

2007-04-12 15:00:03 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-04-12 12:57:33 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-04-12 12:19:20 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-04-12 10:38:03 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/tcp/gstmultifdsink.c:
          log tweaking
          Original commit message from CVS:
          log tweaking

2007-04-12 10:03:22 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
          Original commit message from CVS:
          * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
          (gst_gdp_pay_sink_event):
          Make sure we set the IN_CAPS flag correctly.
          * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
          Get the IN_CAPS flag before we call functions that mess with the flags.

2007-04-10 20:37:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2007-04-10 20:25:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/gdp/gstgdpdepay.c:
        * gst/gdp/gstgdppay.c:
          adding debugging
          Original commit message from CVS:
          adding debugging

2007-04-10 11:23:18 +0000  Christian Schaller <uraeus@gnome.org>

        * common:
        * gst-plugins-base.spec.in:
          update spec file for RTP changes
          Original commit message from CVS:
          update spec file for RTP changes

2007-04-06 12:58:06 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-04-06 09:56:18 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-04-05 15:44:40 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-04-05 10:27:06 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

          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.

2007-04-04 02:45:03 +0000  David Schleef <ds@schleef.org>

          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.

2007-04-03 11:10:52 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

          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.

2007-03-30 17:05:23 +0000  Sebastian Dröge <slomo@circular-chaos.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.

2007-03-30 15:00:49 +0000  Tim-Philipp Müller <tim@centricular.net>

          docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
          Original commit message from CVS:
          * docs/libs/gst-plugins-base-libs-docs.sgml:
          gstreamer-plugins-base.pc doesn't exist, it's
          gstreamer-plugins-base-0.10.pc.

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.

2007-03-27 10:17:16 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-03-27 09:37:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/pipelines/vorbisdec.c:
          commit new file
          Original commit message from CVS:
          commit new file

2007-03-26 22:38:19 +0000  René Stadler <mail@renestadler.de>

          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).

2007-03-26 20:56:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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.

2007-03-26 11:44:07 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-03-23 15:43:24 +0000  Christian Schaller <uraeus@gnome.org>

        * gst-plugins-base.spec.in:
          update spec file
          Original commit message from CVS:
          update spec file

2007-03-23 12:32:33 +0000  Michael Smith <msmith@xiph.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.

2007-03-22 17:43:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-03-22 14:37:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
          Original commit message from CVS:
          * gst-libs/gst/app/Makefile.am:
          Use GST_ALL_LDFLAGS, which actually exists, but maybe David
          can confirm that was what he wanted.

2007-03-22 09:26:02 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-03-21 11:03:23 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
          Original commit message from CVS:
          * gst/playback/gstplaybin.c: (add_sink):
          Use GST_PTR_FORMAT to log caps.

2007-03-21 10:23:11 +0000  Young-Ho Cha <ganadist@chollian.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.

2007-03-20 11:49:55 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-03-19 10:52:50 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-03-18 03:14:01 +0000  David Schleef <ds@schleef.org>

          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.

2007-03-16 17:29:09 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-03-16 16:42:23 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-03-15 10:52:21 +0000  Michael Smith <msmith@xiph.org>

          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.

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.

2007-03-14 15:05:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/gdp/gstgdpdepay.c:
          add buffer logging
          Original commit message from CVS:
          add buffer logging

2007-03-14 14:48:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2007-03-14 14:09:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/audioresample/gstaudioresample.c:
          add debugging and reformat docs
          Original commit message from CVS:
          add debugging and reformat docs

2007-03-12 23:29:07 +0000  Michael Smith <msmith@xiph.org>

          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...

2007-03-12 15:50:35 +0000  Alex Lancaster <alexlan@fedoraproject.org>

        * 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.

2007-03-10 11:21:08 +0000  Tim-Philipp Müller <tim@centricular.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).

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).

2007-03-09 17:33:17 +0000  Tim-Philipp Müller <tim@centricular.net>

          configure.ac: We require core CVS for the new buffer metadata copy functions.
          Original commit message from CVS:
          * configure.ac:
          We require core CVS for the new buffer metadata copy functions.

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.

2007-03-09 13:05:04 +0000  Michael Smith <msmith@xiph.org>

          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.

2007-03-09 12:22:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2007-03-09 11:23:32 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-03-09 10:49:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-03-07 17:15:57 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-03-07 17:12:54 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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

2007-03-07 17:05:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/tcp/gstmultifdsink.c:
          update docs
          Original commit message from CVS:
          update docs

2007-03-07 16:56:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Back to CVS
          Original commit message from CVS:
          * configure.ac:
          Back to CVS

=== release 0.10.12 ===

2007-03-07 16:46:51 +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:
        * win32/common/config.h:
          Release 0.10.12
          Original commit message from CVS:
          Release 0.10.12

2007-03-07 15:35:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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

2007-03-06 12:31:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Bump version to 0.10.11.4 pre-release
          Original commit message from CVS:
          * configure.ac:
          Bump version to 0.10.11.4 pre-release

2007-03-06 12:10:08 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_async_play):
          Fix regression that made GStreamer skip the first samples of audio.
          Fixes #414684.

2007-03-05 11:21:13 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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

2007-03-05 09:35:29 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          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.

2007-03-05 09:27:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-03-04 23:41:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/libs/.gitignore:
          moap ignore
          Original commit message from CVS:
          moap ignore

2007-03-04 23:41:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * win32/common/config.h:
        * win32/common/libgstutils.def:
          update defs
          Original commit message from CVS:
          update defs

2007-03-04 23:39:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

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.

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>

        * tests/check/generic/states.c:
          plug test leak
          Original commit message from CVS:
          plug test leak

2007-02-28 12:44:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2007-02-28 12:08:27 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2007-02-28 11:58:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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.

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.

2007-02-24 20:12:49 +0000  Loïc Minier <lool+gnome@via.ecp.fr>

          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).

2007-02-23 18:49:29 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-02-23 13:10:50 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-02-22 12:57:47 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-02-22 11:04:10 +0000  Andy Wingo <wingo@pobox.com>

        * ChangeLog:
        * gst-libs/gst/audio/gstbaseaudiosink.c:
          gst-libs/gst/audio/gstbaseaudiosink.c
          Original commit message from CVS:
          2007-02-22  Andy Wingo  <wingo@pobox.com>
          * gst-libs/gst/audio/gstbaseaudiosink.c
          (gst_base_audio_sink_init): Disable pull-mode activation until we
          figure out how to make audio sinks go to PLAYING.

2007-02-22 09:04:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
          Original commit message from CVS:
          * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
          (double_hq), (audio_convert_get_func_index),
          (audio_convert_prepare_context), (audio_convert_convert):
          * gst/audioconvert/audioconvert.h:
          * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
          (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
          * gst/audioconvert/gstchannelmix.h:
          * tests/check/elements/audioconvert.c: (GST_START_TEST):
          Add float as an intermediate format, as well as float mixing. Enable
          test that was failing before. Fixes #339837

2007-02-21 16:12:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-02-21 15:36:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-02-21 13:55:54 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2007-02-21 13:08:51 +0000  Stefan Kost <ensonic@users.sourceforge.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.

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).

2007-02-16 10:19:45 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/audio.c: Fix documentation.
          Original commit message from CVS:
          * gst-libs/gst/audio/audio.c:
          Fix documentation.

2007-02-16 10:15:46 +0000  Yves Lefebvre <ivanohe@abacom.com>

          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.

2007-02-15 15:17:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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)

2007-02-15 12:07:57 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-02-15 12:06:25 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosink.c:
          (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
          (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
          (gst_base_audio_sink_async_play),
          (gst_base_audio_sink_change_state):
          Answer latency query.
          Use configured latency when syncing.
          Fix clock slaving.
          * gst-libs/gst/audio/gstbaseaudiosrc.c:
          (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
          (gst_base_audio_src_query), (gst_base_audio_src_change_state):
          Fix possible memleak.
          Implement latency query.
          Small cleanups.

2007-02-15 11:59:41 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
          Original commit message from CVS:
          * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
          Ignore errors in reset, these are not fatal. They also grab the element
          lock which is already taking when this function is called. Fixes
          #405451.

2007-02-13 13:50:56 +0000  Christian Schaller <uraeus@gnome.org>

        * gst-plugins-base.spec.in:
          add header file for easy codec install
          Original commit message from CVS:
          add header file for easy codec install

2007-02-13 10:24:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-02-13 09:12:11 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2007-02-12 20:42:23 +0000  Stefan Kost <ensonic@users.sourceforge.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.

2007-02-12 11:01:04 +0000  René Stadler <mail@renestadler.de>

          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).

2007-02-12 10:33:40 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-02-10 19:27:48 +0000  Sébastien Moutte <sebastien@moutte.net>

          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.

2007-02-10 18:19:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-02-09 13:28:01 +0000  Tim-Philipp Müller <tim@centricular.net>

        * 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).

2007-02-09 09:58:28 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-02-08 15:43:26 +0000  Julien Puydt <julien.puydt@laposte.net>

          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.

2007-02-07 13:05:01 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-02-06 09:42:05 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
          Original commit message from CVS:
          * gst-libs/gst/audio/gstaudiofilter.c:
          (gst_audio_filter_class_init), (gst_audio_filter_change_state):
          Clear our formats structure and free the caps contained in it when
          shutting down.

2007-02-05 18:39:51 +0000  Andy Wingo <wingo@pobox.com>

        * 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).

2007-02-04 16:23:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-02-04 14:11:51 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-02-03 20:19:35 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-02-03 14:26:54 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-02-03 13:59:27 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-02-02 14:44:29 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-02-02 11:21:48 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-02-02 09:48:53 +0000  Stefan Kost <ensonic@users.sourceforge.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)

2007-02-01 18:50:08 +0000  Holger Wansing <linux@wansing-online.de>

          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).

2007-02-01 17:52:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

          ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
          Original commit message from CVS:
          reviewed by: Wim Taymans <wim@fluendo.com>
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
          (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
          Use newly added GstCollectPads API to free the allocated resources in
          the GstOggPad structures (#402393).

2007-01-31 15:58:53 +0000  Jan Schmidt <thaytan@mad.scientist.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

2007-01-30 19:19:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
          (gst_play_base_bin_get_streaminfo_value_array):
          Take some locks and make a copy of the streaminfo value array we
          maintain while holding the lock, so that the application can
          retrieve the stream-info as a value array in a thread-safe way.

2007-01-30 11:29:17 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
          Original commit message from CVS:
          * gst/audioconvert/gstaudioconvert.c:
          Don't fail on 0 sized buffers. Fixes #396835.

2007-01-29 21:13:07 +0000  David Schleef <ds@schleef.org>

          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.

2007-01-29 18:14:25 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

2007-01-25 16:02:41 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (remove_source):
          Don't try to disconnect a signal from a finalized object.

2007-01-25 14:29:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
          Original commit message from CVS:
          * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
          Cast lock macro parameters to make sure we're actually accessing the
          lock member at the right class level. Free list itself in _dispose()
          as well and NULL it in case dispose gets called multiple times.

2007-01-25 14:02:37 +0000  Edward Hervey <bilboed@bilboed.com>

          gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
          Original commit message from CVS:
          * gst/playback/gstdecodebin2.c:
          (gst_decode_bin_dispose),(gst_decode_bin_finalize):
          Free GstDecodeGroups no longer used.
          (gst_decode_group_expose):
          Don't unlock too many times !
          (deactivate_free_recursive):
          Free iterator once we're done with it.
          Fix for recursively deactivating elements (stop at ghostpads).

2007-01-25 12:24:18 +0000  Tim-Philipp Müller <tim@centricular.net>

          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().

2007-01-25 12:06:59 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-25 11:48:10 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2007-01-24 12:51:20 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
          Original commit message from CVS:
          * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
          (gst_text_overlay_text_event):
          Don't unnecessarily ref (and then leak) upstream events if the text
          pad is not linked. Fixes #399948.
          * tests/check/gst-plugins-base.supp:
          Add suppression for pango on edgy/x86 for textoverlay test.

2007-01-24 12:10:56 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2007-01-23 18:39:45 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
          Original commit message from CVS:
          * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
          Error out properly if we get an error from libogg while reading the
          BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).

2007-01-23 17:49:29 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-01-22 10:27:26 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-22 09:23:01 +0000  Tim-Philipp Müller <tim@centricular.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).

2007-01-19 19:09:05 +0000  Tim-Philipp Müller <tim@centricular.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).

2007-01-19 18:47:30 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-19 12:23:06 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-19 11:31:50 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-01-18 16:23:35 +0000  Tim-Philipp Müller <tim@centricular.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.

2007-01-17 16:11:14 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-01-16 19:37:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-01-16 12:17:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-01-16 11:41:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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).

2007-01-15 13:58:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2007-01-15 11:30:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-12 21:19:35 +0000  Andy Wingo <wingo@pobox.com>

        * 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.

2007-01-12 17:43:40 +0000  Christian Schaller <uraeus@gnome.org>

        * gst-plugins-base.spec.in:
          add latest files
          Original commit message from CVS:
          add latest files

2007-01-12 12:47:29 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-12 09:45:23 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: Fix comment.
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c:
          Fix comment.

2007-01-11 13:12:17 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
          Original commit message from CVS:
          * gst/playback/gstplaybin.c: (post_missing_element_message),
          (gen_video_element), (gen_text_element), (gen_audio_element),
          (gen_vis_element):
          Post missing-plugin messages also when we error out because
          converters, textoverlay or auto*sinks are missing (#161922).

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.

2007-01-09 14:20:08 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-09 13:35:08 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

2007-01-08 14:01:23 +0000  Günter Thelen <daedalus.inc@gmx.net>

          gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
          Original commit message from CVS:
          Patch by: Günter Thelen  <daedalus dot inc at gmx net>
          * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
          (plugin_init):
          Add typefinder for flac-in-ogg in conformance with the ogg-mapping
          on flac.sf.net (there appear to be other versions of the first
          ogg page in the wild) (#391365).

2007-01-08 13:32:32 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-01-08 12:30:03 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2007-01-06 17:28:40 +0000  Andy Wingo <wingo@pobox.com>

        * 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>

          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.

2007-01-05 18:57:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2007-01-05 16:02:50 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2007-01-05 12:19:34 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-05 12:13:24 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2007-01-04 12:49:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
          Original commit message from CVS:
          * configure.ac:
          split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
          so that GST_BASE_CFLAGS can go inbetween them, making sure
          we use uninstalled gst-libs headers
          * docs/libs/Makefile.am:
          * ext/alsa/Makefile.am:
          * ext/cdparanoia/Makefile.am:
          * ext/gnomevfs/Makefile.am:
          * ext/libvisual/Makefile.am:
          * ext/ogg/Makefile.am:
          * ext/theora/Makefile.am:
          * ext/vorbis/Makefile.am:
          * gst-libs/gst/audio/Makefile.am:
          * gst-libs/gst/cdda/Makefile.am:
          * gst-libs/gst/interfaces/Makefile.am:
          * gst-libs/gst/riff/Makefile.am:
          * gst-libs/gst/rtp/Makefile.am:
          * gst-libs/gst/tag/Makefile.am:
          * gst/adder/Makefile.am:
          * gst/audioconvert/Makefile.am:
          * gst/audiorate/Makefile.am:
          * gst/audioresample/Makefile.am:
          * gst/playback/Makefile.am:
          * gst/tcp/Makefile.am:
          * gst/videoscale/Makefile.am:
          * gst/volume/Makefile.am:
          * sys/ximage/Makefile.am:
          * sys/xvimage/Makefile.am:
          * tests/icles/Makefile.am:
          adapt

2007-01-04 11:30:53 +0000  Julien Moutte <julien@moutte.net>

          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).

2007-01-01 18:59:49 +0000  Tim-Philipp Müller <tim@centricular.net>

          Dist design docs.
          Original commit message from CVS:
          * configure.ac:
          * docs/Makefile.am:
          * docs/design/Makefile.am:
          Dist design docs.

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

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)

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

2006-12-20 10:29:58 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-12-20 09:25:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-12-16 13:59:09 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-12-16 12:22:57 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-12-15 10:52:23 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-12-15 00:20:37 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          add doap file
          Original commit message from CVS:
          * Makefile.am:
          * gst-plugins-base.doap:
          * gst-plugins-base.spec.in:
          add doap file

2006-12-09 15:12:38 +0000  Jens Granseuer <jensgr@gmx.net>

          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.

2006-12-07 02:38:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: Bump version nano - back to CVS.
          Original commit message from CVS:
          * configure.ac:
          Bump version nano - back to CVS.

=== release 0.10.11 ===

2006-12-07 02:30:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          configure.ac: releasing 0.10.11, "Dumb things"
          Original commit message from CVS:
          === release 0.10.11 ===
          2006-12-06  Jan Schmidt <thaytan@mad.scientist.com>
          * configure.ac:
          releasing 0.10.11, "Dumb things"

2006-12-05 12:44:00 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
          Original commit message from CVS:
          * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
          (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
          Handle the case where an element has multiple pads with
          unfixed caps as well as still possibly producing more dynamic
          pads by storing each case as a distinct entry in the dynamic list.
          Fixes #38223 again.

2006-12-04 13:02:37 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
          Original commit message from CVS:
          * gst/playback/gstdecodebin.c: (close_pad_link):
          Fix #382223, add more dynamic caps handling.

2006-12-01 11:35:57 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

        * po/.gitignore:
          Ignore all pot files
          Original commit message from CVS:
          Ignore all pot files

2006-12-01 10:36:50 +0000  Michael Smith <msmith@xiph.org>

          gst/audiorate/gstaudiorate.c: Delete bad debug code.
          Original commit message from CVS:
          * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
          Delete bad debug code.
          Fixes #381219

2006-12-01 10:27:54 +0000  Sergey Scobich <sergey.scobich@gmail.com>

          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

2006-11-30 23:46:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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

2006-11-30 12:50:42 +0000  Michael Smith <msmith@xiph.org>

          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.

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

2006-11-24 08:35:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2006-11-23 11:07:23 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
          Original commit message from CVS:
          * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
          (gst_ogg_demux_chain):
          Don't just ignore return values from _pad_push().
          Small debug improvements.

2006-11-23 11:02:11 +0000  Michael Smith <msmith@xiph.org>

          ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
          If our incoming buffer is marked as DISCONT, then increment the page
          number (so that the discontinuity is marked in the final ogg
          bitstream) and flush the previous page.

2006-11-22 14:34:03 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-11-21 18:39:34 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-11-21 09:29:56 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-11-21 08:17:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-11-20 16:44:28 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-11-20 15:01:09 +0000  Wim Taymans <wim.taymans@gmail.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.

2006-11-20 12:20:39 +0000  Tim-Philipp Müller <tim@centricular.net>

          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

2006-11-19 17:07:34 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-11-16 14:35:30 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-11-16 12:55:08 +0000  Michael Smith <msmith@xiph.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.

2006-11-15 10:05:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-11-14 23:34:19 +0000  David Schleef <ds@schleef.org>

          configure.ac: Bump liboil requirement to 0.3.8.
          Original commit message from CVS:
          * configure.ac:
          Bump liboil requirement to 0.3.8.
          * gst-libs/gst/riff/riff-media.c:
          Add Dirac fourcc.
          * gst/videoscale/vs_image.h:
          * gst/videoscale/vs_scanline.h:
          Use liboil's stdint.h.
          * gst/videotestsrc/videotestsrc.c:
          Remove liboil related ifdef's, since they aren't needed now, and
          won't work with future versions.

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.

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).

2006-11-10 19:20:21 +0000  Sergey Scobich <sergery.scobich@gmail.com>

          win32/common/libgsttag.def: Add missing symbol (#366492).
          Original commit message from CVS:
          Patch by: Sergey Scobich  <sergery.scobich at gmail com>
          * win32/common/libgsttag.def:
          Add missing symbol (#366492).

2006-11-10 00:52:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-11-09 00:50:00 +0000  Christian Schaller <uraeus@gnome.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.

2006-11-07 07:22:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-11-06 18:24:59 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-11-04 07:25:58 +0000  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.

2006-11-03 15:44:31 +0000  Michael Smith <msmith@xiph.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.

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).

2006-11-01 14:08:31 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-31 15:05:33 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-10-31 14:19:07 +0000  Michael Smith <msmith@xiph.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.

2006-10-31 10:49:19 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-10-28 17:22:57 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-10-28 16:00:51 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
          Original commit message from CVS:
          * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
          Another typo fix (#366212).

2006-10-27 17:13:35 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-10-27 16:46:15 +0000  Wim Taymans <wim.taymans@gmail.com>

        * ChangeLog:
          I'm too lazy to comment this
          Original commit message from CVS:
          *** empty log message ***

2006-10-27 16:45:30 +0000  Mark Nauwelaerts <manauw@skynet.be>

          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.

2006-10-27 11:57:18 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-10-26 10:49:00 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

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.

2006-10-23 11:47:04 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-21 16:39:54 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-10-20 17:02:19 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-20 13:56:55 +0000  Stefan Kost <ensonic@users.sf.net>

          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)

2006-10-20 13:54:19 +0000  Michael Smith <msmith@xiph.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.

2006-10-20 12:31:02 +0000  Michael Smith <msmith@xiph.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.

2006-10-19 14:09:30 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
          (plugin_init):
          Typefind mmsh header data packet to application/x-mmsh (#362625).

2006-10-19 09:17:48 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-19 09:00:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-18 18:40:12 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-18 15:13:59 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
          Original commit message from CVS:
          * gst/subparse/gstsubparse.c:
          (gst_sub_parse_data_format_autodetect):
          Don't require subrip (.srt) files to start with a chunk number of 1.

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.

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.

2006-10-18 09:46:35 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-18 09:31:49 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-10-17 17:17:16 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

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).

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).

2006-10-09 15:01:30 +0000  Edgard Lima <edgard.lima@indt.org.br>

        * 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.

2006-10-08 16:59:31 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-07 18:35:39 +0000  Julien Moutte <julien@moutte.net>

          ext/pango/gsttextoverlay.c: Useless goto.
          Original commit message from CVS:
          2006-10-07  Julien MOUTTE  <julien@moutte.net>
          * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
          Useless goto.
          * tests/examples/seek/seek.c: (do_seek),
          (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
          seek example to experiment with rates != 1.0 (reverse playback
          !)

2006-10-06 19:20:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
          Original commit message from CVS:
          * gst-libs/gst/interfaces/xoverlay.c:
          Unref message in doc-example (spotted by Robert McQueen)

2006-10-06 17:04:05 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/typefind/gsttypefindfunctions.c: printf fix.
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
          (mpeg1_parse_header), (mpeg1_sys_type_find):
          printf fix.

2006-10-06 14:37:43 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/: Activate dynamic pads before adding them to the element.
          Original commit message from CVS:
          * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
          (close_pad_link):
          * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
          Activate dynamic pads before adding them to the element.

2006-10-06 14:04:53 +0000  Michael Smith <msmith@xiph.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.

2006-10-06 13:34:46 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-10-06 12:57:10 +0000  Michael Smith <msmith@xiph.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.

2006-10-06 10:04:14 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-10-05 15:55:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-10-04 13:18:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-10-04 13:15:36 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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.

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.

2006-10-03 10:36:38 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2006-09-29 13:46:45 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (setup_subtitle),
          (gen_source_element):
          Handle invalid URIs a bit more gracefully.

2006-09-29 12:54:28 +0000  Tim-Philipp Müller <tim@centricular.net>

          tests/check/pipelines/oggmux.c: Remove obsolete comment.
          Original commit message from CVS:
          * tests/check/pipelines/oggmux.c:
          Remove obsolete comment.

2006-09-29 10:43:05 +0000  James Doc Livingston <doclivingston@gmail.com>

          ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
          (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
          (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
          (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
          (gst_ogg_mux_collected):
          Commit patch from James "Doc" Livingston, adds proper EOS handling
          in oggmux. GStreamer can, for the first time ever, create a valid
          Ogg file! Yay!
          * tests/check/pipelines/oggmux.c: (check_chain_final_state),
          (oggmux_suite):
          Reenable tests now that they pass.

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?)

2006-09-28 10:49:56 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-09-28 09:41:20 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

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-plugins-base.spec.in:
          add new header file to spec
          Original commit message from CVS:
          add new header file to spec

2006-09-27 12:55:45 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-09-27 11:58:17 +0000  Jonathan Matthew <jonathan@kaolin.wh9.net>

          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

2006-09-27 11:31:43 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

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>

        * ChangeLog:
          ChangeLog surgery for missing bug-number
          Original commit message from CVS:
          ChangeLog surgery for missing bug-number

2006-09-25 11:28:15 +0000  Peter Kjellerstedt <pkj@axis.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).

2006-09-25 10:21:31 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-09-23 15:24:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-09-23 15:02:51 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-09-23 14:30:53 +0000  Young-Ho Cha <ganadist@chollian.net>

          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).

2006-09-23 13:32:07 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2006-09-23 08:53:30 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (gen_preroll_element):
          Improve buffering a bit by avoiding a deadlock because we cannot assume
          the underrun is always called.

2006-09-23 08:51:14 +0000  Young-Ho Cha <ganadist@chollian.net>

          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.

2006-09-22 14:50:01 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2006-09-22 09:52:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/videorate/gstvideorate.c:
          update docs
          Original commit message from CVS:
          update docs

2006-09-21 13:49:47 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
          Original commit message from CVS:
          * gst-libs/gst/interfaces/videoorientation.c:
          (gst_video_orientation_iface_init),
          (gst_video_orientation_get_hflip),
          (gst_video_orientation_get_vflip),
          (gst_video_orientation_get_hcenter),
          (gst_video_orientation_get_vcenter),
          (gst_video_orientation_set_hflip),
          (gst_video_orientation_set_vflip),
          (gst_video_orientation_set_hcenter),
          (gst_video_orientation_set_vcenter):
          Add since tags to new API docs, ChangeLog surgery (forgot API keyword
          in ChangeLog)

2006-09-21 09:27:47 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

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

2006-09-20 10:42:34 +0000  Tim-Philipp Müller <tim@centricular.net>

        * 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

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().

2006-09-18 15:00:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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).

2006-09-18 14:42:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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()

2006-09-18 14:21:45 +0000  Michael Smith <msmith@xiph.org>

          ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
          Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
          always true, leading to dropping all timestamps.

2006-09-18 11:40:14 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-09-18 10:57:28 +0000  Michael Smith <msmith@xiph.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

2006-09-18 10:18:22 +0000  Michael Smith <msmith@xiph.org>

          gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
          Original commit message from CVS:
          * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
          (gst_multi_fd_sink_class_init):
          Updates, fixes, and typo corrections for multifdsink. No functional
          changes.

2006-09-17 21:58:06 +0000  Michael Smith <msmith@xiph.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.

2006-09-17 20:32:09 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-09-17 20:14:43 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

=== release 0.10.10 ===

2006-09-14 20:09:19 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2006-09-09 16:08:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * configure.ac:
        * win32/common/config.h:
          second prerelease
          Original commit message from CVS:
          second prerelease

2006-09-07 19:01:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
          update bug in changelog
          Original commit message from CVS:
          update bug in changelog

2006-09-07 19:00:33 +0000  Michael Smith <msmith@fluendo.com>

          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'

2006-09-07 15:00:08 +0000  Wim Taymans <wim@fluendo.com>

          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

2006-09-07 14:56:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * configure.ac:
        * win32/common/config.h:
          first prerelease
          Original commit message from CVS:
          first prerelease

2006-09-07 14:56:31 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-09-05 09:12:25 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-09-04 19:04:35 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

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.

2006-09-04 09:13:01 +0000  Alessandro Dessina <alessandro@nnva.org>

          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

2006-09-02 13:20:59 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

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").

2006-09-01 10:07:05 +0000  Tim-Philipp Müller <tim@centricular.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).

2006-08-31 14:37:33 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/adder/gstadder.c: Fix build.
          Original commit message from CVS:
          * gst/adder/gstadder.c: (gst_adder_src_event):
          Fix build.

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).

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.

2006-08-29 10:58:43 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-08-29 10:51:12 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-08-29 10:32:34 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-08-29 08:03:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-08-28 16:17:13 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-08-28 16:09:57 +0000  Wim Taymans <wim.taymans@gmail.com>

        * 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.

2006-08-25 16:39:38 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/tcp/gstmultifdsink.c: Small cleanups.
          Original commit message from CVS:
          * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
          Small cleanups.
          If a buffer is received with no caps, make the buffer metadata
          writable and set the caps, making sure that we don't screw up the
          refcounts.

2006-08-25 16:19:55 +0000  Michael Smith <msmith@xiph.org>

          gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
          Original commit message from CVS:
          * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
          (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
          Fix memory leaks and misleading debug messages, add a couple of
          comments.
          * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
          (gst_multi_fd_sink_render):
          Do not use gst_buffer_make_writable() in a basesink render method,
          as it may incorrectly unref the buffer. Instead, use convoluted
          dance to avoid copying the buffer except when we need to.

2006-08-25 09:54:56 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-08-24 11:18:56 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-08-23 19:37:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/check/elements/.cvsignore: make buildbot happy
          Original commit message from CVS:
          * tests/check/elements/.cvsignore:
          make buildbot happy

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.

2006-08-23 15:27:38 +0000  Wim Taymans <wim.taymans@gmail.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.

2006-08-23 15:11:56 +0000  Tim-Philipp Müller <tim@centricular.net>

          tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
          Original commit message from CVS:
          * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
          Unit test fixes: \377 is more likely to fit into 8 bits than \777;
          actually return return value in taglists_are_equal.

2006-08-23 12:14:20 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
          Original commit message from CVS:
          * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
          Fix crash due to broken bitstream parsing on x86-64: can't make
          any assumptions about sizeof(struct) due to alignment/packing
          differences on different architectures. Fixes #351790.

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.

2006-08-22 15:50:36 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-08-22 15:11:52 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-08-22 08:27:07 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-08-21 19:04:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/gdp/README:
          adding a README
          Original commit message from CVS:
          adding a README

2006-08-21 19:01:41 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          Move GDP plugin to -base from -bad.  Closes #347783.
          Original commit message from CVS:
          * configure.ac:
          * docs/plugins/Makefile.am:
          * docs/plugins/gst-plugins-base-plugins-docs.sgml:
          * docs/plugins/gst-plugins-base-plugins-sections.txt:
          * docs/plugins/inspect/plugin-gdp.xml:
          * gst/gdp/Makefile.am:
          * tests/check/Makefile.am:
          Move GDP plugin to -base from -bad.  Closes #347783.

2006-08-21 18:34:46 +0000  Tim-Philipp Müller <tim@centricular.net>

          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().

2006-08-21 16:39:25 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

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

2006-08-20 13:05:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-08-18 21:21:48 +0000  Tim-Philipp Müller <tim@centricular.net>

        * 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.

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.

2006-08-16 11:38:52 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/alsa/: Avoid setting and using a NULL device name.
          Original commit message from CVS:
          * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
          (gst_alsasink_open):
          * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
          (gst_alsasrc_open):
          Avoid setting and using a NULL device name.
          Print more info when we fail to open a device.

2006-08-16 11:28:57 +0000  Tim-Philipp Müller <tim@centricular.net>

          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().

2006-08-15 19:20:16 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-08-15 17:21:33 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-08-14 17:54:01 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-08-14 11:41:04 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2006-08-10 08:56:22 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
          Original commit message from CVS:
          * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
          Implement SEEKING query in its most basic form, so that we can
          at least check if we're seekable or not (#350655).

2006-08-09 14:42:58 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
          The checks here are not even close to anything that would
          justify MAXIMUM probability, lowering to POSSIBLE until someone
          fixes the checks (case at hand: quicktime redirection files
          might start with 00 00 01 XX and pass the checks here just
          fine, see #350399).

2006-08-08 13:57:29 +0000  Edward Hervey <bilboed@bilboed.com>

          tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
          Original commit message from CVS:
          * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
          I forgot to include the file containing the #define :)
          Now includes "config.h"

2006-08-08 13:45:44 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-08-08 08:41:13 +0000  Sjoerd Simons <sjoerd@luon.net>

          gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
          Original commit message from CVS:
          Patch by: Sjoerd Simons  <sjoerd at luon net>
          * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
          Better detection for multipart/x-mixed-replace: accept leading
          whitespaces before the boundary marker as well (as our very own
          multipartmux used to produce) (#349068).

2006-08-07 08:26:03 +0000  Young-Ho Cha <ganadist@chollian.net>

          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).

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.

2006-08-03 14:16:06 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-08-03 11:15:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-08-03 09:01:25 +0000  Jens Granseuer <jensgr@gmx.net>

          gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
          Original commit message from CVS:
          Patch by: Jens Granseuer  <jensgr at gmx net>
          * gst/typefind/gsttypefindfunctions.c: (plugin_init):
          Add typefinder for Interplay's MVE format (#348973).

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).

2006-07-28 14:14:58 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/adder/gstadder.c: Fix leak in duration query.
          Original commit message from CVS:
          * gst/adder/gstadder.c: (gst_adder_setcaps),
          (gst_adder_query_duration):
          Fix leak in duration query.
          Reflow some docs and notes.

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.

2006-07-28 12:48:21 +0000  Michael Smith <msmith@xiph.org>

          ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
          Original commit message from CVS:
          * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
          (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
          (gst_vorbis_enc_push_buffer),
          (gst_vorbis_enc_buffer_check_discontinuous),
          (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
          * ext/vorbis/vorbisenc.h:
          Handle discontinuities in the input vorbis stream correctly,
          so that the output is properly timestamped (and has good granulepos
          values). Needs some oggmux fixes too.

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

2006-07-26 10:55:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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>

        * 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.

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.

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.

2006-07-20 10:42:21 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-07-20 05:56:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

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).

2006-07-19 12:25:00 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
          Original commit message from CVS:
          * gst-libs/gst/tag/gstid3tag.c:
          Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
          * gst-libs/gst/tag/gsttageditingprivate.h:
          * gst-libs/gst/tag/gstvorbistag.c:
          Some more random const-ifications.

2006-07-18 19:48:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-07-17 14:17:45 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-07-17 13:48:10 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
          Original commit message from CVS:
          * gst-libs/gst/audio/multichannel.c:
          (gst_audio_check_channel_positions),
          (gst_audio_fixate_channel_positions):
          Const-ify two arrays.

2006-07-17 12:33:42 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
          Original commit message from CVS:
          * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
          Fix typo, so that alsasink also advertises 8 channels
          if that's supported (tags: can, worms, open, alsa, ph34r).

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

=== release 0.10.9 ===

2006-07-14 15:51:24 +0000  Jan Schmidt <thaytan@mad.scientist.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"

2006-07-14 14:12:40 +0000  Michael Smith <msmith@xiph.org>

          tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
          Original commit message from CVS:
          * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
          Move a g_cond_signal to earlier to avoid sometimes deadlocking
          (commonly happens when running this test under valgrind) when trying
          to remove the buffer probe.

2006-07-14 10:34:36 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/gdp/Makefile.am:
          build as a plugin, not a lib
          Original commit message from CVS:
          build as a plugin, not a lib

2006-07-13 16:43:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
          Original commit message from CVS:
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
          Fix missing g_unlock from the previous commit

2006-07-13 16:34:04 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
          Original commit message from CVS:
          * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
          (gst_ximagesink_change_state):
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
          (gst_xvimagesink_change_state):
          Implement a locking order to ensure we always take the object lock
          before the x_lock and never vice-versa.

2006-07-13 15:25:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          docs/plugins/: add more plugins and elements to docs
          Original commit message from CVS:
          * docs/plugins/Makefile.am:
          * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
          * docs/plugins/gst-plugins-bad-plugins-sections.txt:
          add more plugins and elements to docs
          * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
          fix segfaults due to wrong g_free
          add example
          * gst/gdp/gstgdppay.c:
          add example

2006-07-13 14:38:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
          Original commit message from CVS:
          * gst/playback/gstdecodebin.c: (find_compatibles):
          Fix a caps leak when linking (#347304)
          * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
          (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
          (gst_ximagesink_change_state):
          * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
          (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
          (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
          (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
          Don't leak shared memory resources. Use the object lock to protect
          against the xcontext disappearing while returning a buffer from the
          pipeline. (#347304)

2006-07-12 14:20:43 +0000  Edward Hervey <bilboed@bilboed.com>

          ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
          Original commit message from CVS:
          * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
          (vorbis_handle_comment_packet):
          gst_tag_list_merge() returns a new object. Take that into account when
          using it. This avoids memleak.
          Revert previous commit which is not needed.

2006-07-12 13:30:20 +0000  Edward Hervey <bilboed@bilboed.com>

          ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
          Original commit message from CVS:
          * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
          Reset the decoder in finalize so that all fields get cleared.

2006-07-12 13:24:19 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
          Original commit message from CVS:
          * gst-libs/gst/audio/gstbaseaudiosrc.c:
          (gst_base_audio_src_set_clock),
          (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
          Don't try to post an error message when setting the clock fails
          as this can happen when adding an element to a bin which will then
          deadlock. Fixes #347296.

2006-07-12 13:04:15 +0000  Edward Hervey <bilboed@bilboed.com>

          ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
          Original commit message from CVS:
          * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
          (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
          (vorbis_handle_type_packet):
          Post tag messages on the bus even if we're not initialized.
          If we're not initialized, we still postpone the event pushing of tags.

2006-07-12 11:28:37 +0000  Wim Taymans <wim.taymans@gmail.com>

          Revert last two changes that broke the freeze.
          Original commit message from CVS:
          * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
          * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
          (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
          Revert last two changes that broke the freeze.

2006-07-12 10:59:55 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
          Original commit message from CVS:
          * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
          basesink calculates silence sample correctly for us.

2006-07-12 10:58:42 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
          Original commit message from CVS:
          * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
          (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
          Calculate correct silence samples so we don't fill our ringbuffer
          with noise.

2006-07-12 10:37:18 +0000  Edward Hervey <bilboed@bilboed.com>

          ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
          Original commit message from CVS:
          * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
          (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
          (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
          * ext/vorbis/vorbisdec.h:
          Delay sending events (newsegment, tags) until the decoder is properly
          initialized.
          Fixes #347295

2006-07-11 22:40:13 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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

2006-07-11 21:04:16 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-07-11 12:03:25 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-07-10 14:59:03 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-07-08 13:48:58 +0000  Andy Wingo <wingo@pobox.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).

2006-07-06 13:12:02 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-07-06 13:04:24 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-07-05 20:11:13 +0000  Sébastien Moutte <sebastien@moutte.net>

          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.

2006-07-04 16:50:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/alsa/gstalsamixertrack.c: Some const-ification.
          Original commit message from CVS:
          * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
          Some const-ification.

2006-07-04 14:06:03 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-07-03 10:43:31 +0000  Edward Hervey <bilboed@bilboed.com>

          ext/vorbis/vorbisdec.c: Remove unused variable.
          Original commit message from CVS:
          * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
          Remove unused variable.

2006-07-02 21:48:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          Makefile.am: include lcov.mak
          Original commit message from CVS:
          * Makefile.am:
          include lcov.mak
          * configure.ac:
          add GCOV_LIBS to GST_LIBS

2006-07-02 11:08:58 +0000  Michael Sheldon <webmaster@mikeasoft.com>

          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.

2006-06-30 12:04:51 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-06-29 12:21:06 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

          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

2006-06-26 13:19:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          gst/tcp/gsttcp.c: fix logging
          Original commit message from CVS:
          * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
          (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
          fix logging

2006-06-23 16:45:50 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-06-23 09:53:09 +0000  Tim-Philipp Müller <tim@centricular.net>

          Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
          Original commit message from CVS:
          * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
          * gst-libs/gst/rtp/gstbasertpdepayload.c:
          * gst-libs/gst/rtp/gstbasertppayload.c:
          * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
          * gst/tcp/gstmultifdsink.c:
          * gst/tcp/gsttcpclientsink.c:
          * gst/tcp/gsttcpclientsrc.c:
          * gst/tcp/gsttcpserversink.c:
          * gst/tcp/gsttcpserversrc.c:
          * gst/videorate/gstvideorate.c:
          * gst/videotestsrc/gstvideotestsrc.c:
          * sys/v4l/gstv4ljpegsrc.c:
          * sys/v4l/gstv4lmjpegsink.c:
          * sys/v4l/gstv4lsrc.c:
          * tests/examples/seek/scrubby.c:
          * tests/examples/seek/seek.c:
          Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).

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.

2006-06-22 12:03:14 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-06-22 11:01:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          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'.

2006-06-22 10:55:05 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-06-22 10:31:22 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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).

2006-06-21 18:39:07 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
          Original commit message from CVS:
          * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
          (gst_text_overlay_video_chain):
          g_markup_escape_text() REALLY doesn't like non-UTF8 input
          and doesn't validate its input either (and neither did
          textoverlay it seems). Let's do that then and fix #345206.

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>

        * gst-plugins-base.spec.in:
          update for latest changes
          Original commit message from CVS:
          update for latest changes

2006-06-19 09:57:50 +0000  Edward Hervey <bilboed@bilboed.com>

          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

2006-06-19 09:08:05 +0000  Philip Jaegenstedt <philip@lysator.liu.se>

          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

2006-06-17 14:18:41 +0000  Tim-Philipp Müller <tim@centricular.net>

        * ChangeLog:
          ChangeLog surgery
          Original commit message from CVS:
          ChangeLog surgery

2006-06-17 14:13:03 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-06-16 15:43:23 +0000  Tim-Philipp Müller <tim@centricular.net>

          tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
          Original commit message from CVS:
          * tests/check/elements/audioresample.c: (test_reuse),
          (audioresample_suite):
          Add test case for bug #342789 fixed below.

2006-06-16 15:17:44 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
          Original commit message from CVS:
          * gst/audioresample/gstaudioresample.c:
          (gst_audioresample_class_init), (gst_audioresample_init),
          (audioresample_start), (audioresample_stop),
          (gst_audioresample_set_property), (gst_audioresample_get_property):
          Implement GstBaseTransform::start and ::stop so that audioresample
          can clear its internal state properly and be reused insted of
          causing non-negotiated errors with playbin under some circumstances
          (#342789).
          * tests/check/elements/audioresample.c: (setup_audioresample),
          (cleanup_audioresample):
          Need to set element state here so that ::start and ::stop are
          called.

2006-06-16 13:59:29 +0000  Young-Ho Cha <ganadist@chollian.net>

          gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
          Original commit message from CVS:
          Patch by: Young-Ho Cha <ganadist at chollian dot net>
          * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
          Parse extra data better, apparently it's right behind
          the normal strf header size. Fixes #343500.

2006-06-16 11:04:21 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
          Original commit message from CVS:
          * ext/alsa/gstalsasink.c: (set_hwparams):
          If we fail to set the buffer_time and period_time alsa
          parameters, post a warning and leave alsa select a
          default instead of failing. Fixes #342085

2006-06-16 10:30:25 +0000  Tim-Philipp Müller <tim@centricular.net>

        * ChangeLog:
          ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
          Original commit message from CVS:
          ChangeLog surgery (it wouldn't have crashed, just shown bogus values)

2006-06-16 10:20:10 +0000  Tim-Philipp Müller <tim@centricular.net>

          Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
          Original commit message from CVS:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          * gst-libs/gst/cdda/gstcddabasesrc.h:
          Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
          out in the header file and shouldn't be listed in the docs.
          * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
          Fix it so that it doesn't crash in the debug statement.

2006-06-16 10:02:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          docs/libs/: add remaining symbols into correct setions
          Original commit message from CVS:
          * docs/libs/Makefile.am:
          * docs/libs/gst-plugins-base-libs-docs.sgml:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          * docs/libs/gst-plugins-base-libs.types:
          add remaining symbols into correct setions
          * gst-libs/gst/audio/gstringbuffer.c:
          fix incomplete docs
          * gst-libs/gst/audio/gstringbuffer.h:
          comment out not yet implemented function
          * gst-libs/gst/floatcast/floatcast.h:
          * gst-libs/gst/netbuffer/gstnetbuffer.c:
          add short descriptions
          * gst-libs/gst/interfaces/propertyprobe.c:
          fix return value docs
          * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
          simplify debug logging
          * gst-libs/gst/riff/riff-read.h:
          sync function prototype and docs
          * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
          remove left over symbol

2006-06-16 09:56:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-06-15 15:27:49 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-06-14 14:49:33 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-06-14 14:34:28 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-06-14 09:13:54 +0000  Alessandro Decina <alessandro@nnva.org>

          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.

2006-06-14 08:17:45 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-06-12 12:44:38 +0000  Young-Ho Cha <ganadist@chollian.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).

2006-06-11 20:41:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          po/POTFILES.in: add missing files containing translatable strings
          Original commit message from CVS:
          * po/POTFILES.in:
          add missing files containing translatable strings

2006-06-11 19:55:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-06-11 19:44:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          ignore more
          Original commit message from CVS:
          * docs/libs/.cvsignore:
          * tests/check/elements/.cvsignore:
          * tests/check/libs/.cvsignore:
          ignore more

2006-06-11 18:33:27 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
          Original commit message from CVS:
          * docs/libs/Makefile.am:
          also commiting the changed Makefile.am (added more libs to the
          doc-build)

2006-06-11 17:08:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          docs/libs/: first batch of reordering things, add index & hierarchy
          Original commit message from CVS:
          * docs/libs/gst-plugins-base-libs-docs.sgml:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          * docs/libs/gst-plugins-base-libs.types:
          first batch of reordering things, add index & hierarchy

2006-06-11 14:08:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * common:
        * ext/alsa/Makefile.am:
        * ext/cdparanoia/Makefile.am:
        * ext/gnomevfs/Makefile.am:
        * ext/libvisual/Makefile.am:
        * ext/ogg/Makefile.am:
        * ext/pango/Makefile.am:
        * ext/theora/Makefile.am:
        * ext/vorbis/Makefile.am:
        * sys/v4l/Makefile.am:
        * sys/ximage/Makefile.am:
        * sys/xvimage/Makefile.am:
          further clean up build
          Original commit message from CVS:
          further clean up build

2006-06-11 12:14:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-06-11 12:10:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * common:
        * win32/common/config.h:
          update to cvs
          Original commit message from CVS:
          update to cvs

2006-06-10 18:52:03 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
          Original commit message from CVS:
          * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
          Add support for burn:// URIs (#343385); const-ify things a bit,
          use G_N_ELEMENTS instead of hard-coded array size.

2006-06-10 18:25:07 +0000  Young-Ho Cha <ganadist@chollian.net>

          gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
          Original commit message from CVS:
          Patch by: Young-Ho Cha  <ganadist at chollian net>
          * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
          Fix up broken entities before passing them to libxml *sigh*.
          (#343303).

2006-06-09 18:52:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * configure.ac:
          back to trunk
          Original commit message from CVS:
          back to trunk

=== release 0.10.8 ===

2006-06-09 18:49:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playbin.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * win32/common/config.h:
          releasing 0.10.8
          Original commit message from CVS:
          releasing 0.10.8

2006-06-07 11:04:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-06-07 11:03:03 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-06-06 16:26:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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.

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.

2006-06-02 16:45:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2006-06-02 15:06:59 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-06-02 14:23:34 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-06-02 14:19:18 +0000  Michael Smith <msmith@xiph.org>

          gst/audioconvert/audioconvert.c: More correct float->int conversion.
          Original commit message from CVS:
          * gst/audioconvert/audioconvert.c: (float):
          More correct float->int conversion.

2006-06-02 14:07:42 +0000  Michael Smith <msmith@xiph.org>

          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.

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/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

2006-06-01 23:04:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

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

2006-05-30 20:33:59 +0000  Sébastien Moutte <sebastien@moutte.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

2006-05-30 19:00:39 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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

2006-05-30 14:31:43 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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.

2006-05-30 11:45:52 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-05-29 17:19:48 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

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).

2006-05-29 11:04:48 +0000  Michael Smith <msmith@xiph.org>

          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.

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.

2006-05-26 15:52:23 +0000  Wim Taymans <wim.taymans@gmail.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.

2006-05-26 11:17:57 +0000  Jan Schmidt <thaytan@mad.scientist.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.

2006-05-26 10:18:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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).

2006-05-24 08:34:53 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-05-23 20:38:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-05-23 15:18:40 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2006-05-22 11:42:03 +0000  Peter Kjellerstedt <pkj@axis.com>

          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).

2006-05-19 17:57:56 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

          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()

2006-05-19 15:00:43 +0000  Tim-Philipp Müller <tim@centricular.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.

2006-05-19 13:37:55 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

2006-05-18 19:21:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

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>

2006-05-18 10:05:23 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
          Original commit message from CVS:
          * gst/subparse/samiparse.c: (handle_start_font):
          Don't ignore return value of strtol (++compiler_happiness).

2006-05-17 17:49:10 +0000  Young-Ho Cha <ganadist@chollian.net>

          gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
          Original commit message from CVS:
          Patch by: Young-Ho Cha  <ganadist chollian net>
          * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
          (gst_sub_parse_class_init), (gst_sub_parse_init),
          (gst_sub_parse_set_property), (gst_sub_parse_get_property),
          (convert_encoding):
          * gst/subparse/gstsubparse.h:
          Add 'encoding' property (#341681).
          * gst/subparse/samiparse.c: (characters_sami):
          Output is pango markup, so we need to escape text
          between tags (#342143).

2006-05-16 17:34:14 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
          Original commit message from CVS:
          * gst-libs/gst/audio/multichannel.c:
          (gst_audio_check_channel_positions):
          It's okay to have caps with channels=1 and a channel position
          different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
          (deinterleavers might want to keep the position in the caps,
          so that they can be re-interleaved again properly later).
          Leave check for unexpected 2-channel layouts intact for now.

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.

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.

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.

2006-05-15 15:31:30 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-05-15 15:01:08 +0000  Wim Taymans <wim.taymans@gmail.com>

          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

2006-05-15 14:19:35 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-05-15 12:18:13 +0000  Tim-Philipp Müller <tim@centricular.net>

          configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
          Original commit message from CVS:
          * configure.ac:
          Build video4linux plugin even if there's no XVIDEO, just
          without implementing the GstXOverlay interface (#334002).

2006-05-15 10:17:04 +0000  Tim-Philipp Müller <tim@centricular.net>

          Add tentative support for libvisual-0.4 (#336881).
          Original commit message from CVS:
          * configure.ac:
          * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
          (plugin_init):
          Add tentative support for libvisual-0.4 (#336881).

2006-05-15 09:41:03 +0000  Young-Ho Cha <ganadist@chollian.net>

          gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
          Original commit message from CVS:
          Patch by: Young-Ho Cha <ganadist at chollian net>
          * gst/subparse/samiparse.c: (handle_start_font):
          Need to map "silver" colour explicitly (#169936).

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).

2006-05-14 21:18:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * win32/common/config.h:
          update config.h
          Original commit message from CVS:
          update config.h

2006-05-14 21:18:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ext/ogg/README:
          fix mistakes in README
          Original commit message from CVS:
          fix mistakes in README

2006-05-14 18:15:17 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-05-14 16:05:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * configure.ac:
          Back to CVS
          Original commit message from CVS:
          Back to CVS

=== release 0.10.7 ===

2006-05-14 16:00:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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"

2006-05-14 15:55:16 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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

2006-05-12 22:22:37 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

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)

2006-05-12 16:56:52 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
          Original commit message from CVS:
          * gst-libs/gst/tag/gstid3tag.c:
          Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).

2006-05-12 10:39:08 +0000  Sébastien Moutte <sebastien@moutte.net>

          win32/MANIFEST: update win32 files listing
          Original commit message from CVS:
          * win32/MANIFEST:
          update win32 files listing

2006-05-11 21:47:01 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/elements/multifdsink.c:
          disable failing check on gentoo64
          Original commit message from CVS:
          disable failing check on gentoo64

2006-05-11 21:35:44 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/elements/multifdsink.c:
          disable failing check on gentoo64
          Original commit message from CVS:
          disable failing check on gentoo64

2006-05-11 21:20:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/elements/multifdsink.c:
          macros show the correct line
          Original commit message from CVS:
          macros show the correct line

2006-05-11 21:04:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/elements/multifdsink.c:
          macros show the correct line
          Original commit message from CVS:
          macros show the correct line

2006-05-11 21:01:05 +0000  Sjoerd Simons <sjoerd@luon.net>

          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

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).

2006-05-11 16:04:28 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-05-11 12:34:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/tcp/gstmultifdsink.c:
          more commenting
          Original commit message from CVS:
          more commenting

2006-05-11 11:40:39 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-05-11 10:33:46 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-05-11 10:14:20 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-05-10 18:31:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/elements/multifdsink.c:
          set caps and plug leaks
          Original commit message from CVS:
          set caps and plug leaks

2006-05-10 18:16:23 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-05-10 16:34:06 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-05-10 15:16:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/tcp/gstmultifdsink.c:
          DEBUG_FUNCPTR'ing
          Original commit message from CVS:
          DEBUG_FUNCPTR'ing

2006-05-10 15:14:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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>

          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.

2006-05-09 17:40:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          configure.ac: Require core CVS for the new newsegment stuff.
          Original commit message from CVS:
          * configure.ac:
          Require core CVS for the new newsegment stuff.

2006-05-09 17:30:48 +0000  Sjoerd Simons <sjoerd@luon.net>

          gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
          Original commit message from CVS:
          Patch by: Sjoerd Simons  <sjoerd at luon net>
          * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
          Register nick for enum value (#341160).

2006-05-09 16:46:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-05-09 16:14:49 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-05-09 11:59:13 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-05-09 11:37:22 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-05-09 11:31:47 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-05-09 11:21:24 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-05-08 15:51:15 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * docs/libs/tmpl/gstcolorbalance.sgml:
        * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
        * gst/tcp/gstmultifdsink.c:
        * gst/videoscale/gstvideoscale.c:
          doc reparagraphing and DEBUG_FUNCPTRing
          Original commit message from CVS:
          doc reparagraphing and DEBUG_FUNCPTRing

2006-05-08 11:51:43 +0000  Edward Hervey <bilboed@bilboed.com>

          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

2006-05-07 17:39:04 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
          Original commit message from CVS:
          * tests/check/Makefile.am:
          * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
          Disable the adder test, until the build-slaves posses the kindness to
          either like it or to give valid reason for not doing so

2006-05-07 17:25:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

2006-05-07 17:07:03 +0000  Stefan Kost <ensonic@users.sourceforge.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

2006-05-07 16:39:36 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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.

2006-05-05 16:34:15 +0000  Andy Wingo <wingo@pobox.com>

        * 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.

2006-05-05 12:37:35 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
          (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
          Rearrange MPEG system stream detection, fixing some memleaks in the
          process.
          Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
          they clean up their data correctly.
          Remove unused ogganx caps and move the 'is_annodex' check to inside
          the 'is_ogg' if statement.

2006-05-05 11:33:37 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
          Original commit message from CVS:
          * gst/playback/gstdecodebin.c: (cleanup_decodebin):
          Properly remove ghostpads. Fixes #340392

2006-05-04 18:43:58 +0000  David Schleef <ds@schleef.org>

          gst/typefind/gsttypefindfunctions.c:
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c:

2006-05-03 16:32:19 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
          (mpeg_ts_probe_headers), (mpeg_ts_type_find):
          When typefinding an MP3 in push-based mode, don't penalise the
          probability down to 74% when we found 5 valid frames just because we
          can't peek the end of the file.
          Make the probability for detecting MPEG Transport Streams based on the
          number of sequential headers we successfully detected.

2006-05-03 15:52:46 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
          Original commit message from CVS:
          * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
          (vorbis_dec_push), (vorbis_dec_chain):
          Still produce an error when we receive an empty packet.

2006-05-03 15:34:48 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
          Original commit message from CVS:
          * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
          (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
          (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
          Mark buffers with DISCONT after seek and after activating new
          chains.
          * ext/theora/gsttheoradec.h:
          * ext/theora/theoradec.c: (gst_theora_dec_reset),
          (theora_get_query_types), (theora_dec_sink_event),
          (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
          (theora_dec_change_state):
          Fix frame counter.
          Detect and mark DISCONT buffers.
          * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
          (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
          (vorbis_dec_change_state):
          * ext/vorbis/vorbisdec.h:
          Use GstSegment.
          Detect and mark DISCONT buffers.
          Don't crash on 0 sized buffers.

2006-05-03 08:58:13 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
          Original commit message from CVS:
          * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
          (volume_transform_ip):
          Increase "volume" property to 10.0. Fixes #340369.
          Set the process function to NULL when capsnego fails so that
          we properly error out.

2006-05-02 18:15:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
          Original commit message from CVS:
          * gst/playback/gstplaybin.c: (add_sink):
          * gst/playback/test.c: (main):
          * gst/playback/test5.c: (dump_element_stats):
          * gst/playback/test6.c: (main):
          free cpas using gst_caps_unref, don't leak caps-strings

2006-05-02 06:33:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst-libs/gst/rtp/gstbasertppayload.c:
          some RTP debug
          Original commit message from CVS:
          some RTP debug

2006-05-01 19:08:40 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-04-29 16:25:58 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-04-29 13:09:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-04-29 12:10:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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>

          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

2006-04-29 09:48:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ext/ogg/gstoggmux.c:
        * ext/theora/theoraenc.c:
          add another include
          Original commit message from CVS:
          add another include

2006-04-29 01:24:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/subparse/gstssaparse.c:
          atoi() needs stdlib.h
          Original commit message from CVS:
          atoi() needs stdlib.h

2006-04-29 01:18:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2006-04-29 01:10:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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>

2006-04-29 01:04:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2006-04-28 19:46:37 +0000  Stefan Kost <ensonic@users.sourceforge.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

2006-04-28 19:17:06 +0000  Stefan Kost <ensonic@users.sourceforge.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

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.

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.

2006-04-28 14:49:22 +0000  Thomas Vander Stichele <thomas@apestaart.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

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>

        * 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).

2006-04-28 12:15:33 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-04-28 10:58:41 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * configure.ac:
          back to HEAD
          Original commit message from CVS:
          back to HEAD

=== release 0.10.6 ===

2006-04-28 10:53:40 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2006-04-28 10:42:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * win32/MANIFEST:
        * win32/common/config.h:
          dist more win32 files
          Original commit message from CVS:
          dist more win32 files

2006-04-28 10:41:57 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-04-27 00:19:29 +0000  David Schleef <ds@schleef.org>

          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.

2006-04-26 17:20:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * configure.ac:
        * win32/common/config.h:
          new prerelease
          Original commit message from CVS:
          new prerelease

2006-04-26 17:17:39 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-04-26 16:44:20 +0000  Edward Hervey <edward@fluendo.com>

          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.

2006-04-26 13:55:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * configure.ac:
        * win32/common/config.h:
          prerelease
          Original commit message from CVS:
          prerelease

2006-04-22 21:25:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-04-22 21:19:06 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-04-22 21:12:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-04-13 09:26:27 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-04-13 03:55:12 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

          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

2006-04-12 11:04:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * configure.ac:
        * po/af.po:
        * po/az.po:
        * po/cs.po:
        * po/en_GB.po:
        * po/hu.po:
        * po/it.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/uk.po:
        * po/vi.po:
          update libtool versioning
          Original commit message from CVS:
          update libtool versioning

2006-04-12 10:58:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * configure.ac:
        * win32/common/config.h:
          prerelease
          Original commit message from CVS:
          prerelease

2006-04-11 17:31:29 +0000  Antoine Tremblay <hexa00@gmail.com>

          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.

2006-04-11 15:01:51 +0000  Christian Schaller <uraeus@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.

2006-04-11 09:42:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
          Mention bug #336617 closed by recent commit
          Original commit message from CVS:
          Mention bug #336617 closed by recent commit

2006-04-10 20:32:46 +0000  Michael Smith <msmith@xiph.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.

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

2006-04-10 10:29:21 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-04-09 17:14:22 +0000  Sébastien Moutte <sebastien@moutte.net>

          gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
          Original commit message from CVS:
          * gst/audioresample/debug.h:
          replace debug macros with variable number of parameters
          by a simple alias to gstreamer standard debug macros
          (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
          supported by MSVC 6.0 and 7.1)
          * gst/audioresample/resample.h:
          define M_PI and rint for WIN32
          * win32/common/libgstaudio.def:
          * win32/common/libgstriff.def:
          * win32/common/libgsttag.def:
          * win32/common/libgstvideo.def:
          add new exported functions
          * win32/vs6:
          update project files

2006-04-08 21:02:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
          Original commit message from CVS:
          * ext/alsa/gstalsamixeroptions.c:
          (gst_alsa_mixer_options_class_init):
          * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
          * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
          * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
          * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
          * gst-libs/gst/audio/gstaudiofilter.c:
          (gst_audio_filter_class_init):
          * gst-libs/gst/audio/gstaudiosink.c:
          (gst_audioringbuffer_class_init):
          * gst-libs/gst/audio/gstaudiosrc.c:
          (gst_audioringbuffer_class_init):
          * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
          * gst-libs/gst/interfaces/colorbalancechannel.c:
          (gst_color_balance_channel_class_init):
          * gst-libs/gst/interfaces/mixeroptions.c:
          (gst_mixer_options_class_init):
          * gst-libs/gst/interfaces/mixertrack.c:
          (gst_mixer_track_class_init):
          * gst-libs/gst/interfaces/tunerchannel.c:
          (gst_tuner_channel_class_init):
          * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
          * gst-libs/gst/netbuffer/gstnetbuffer.c:
          (gst_netbuffer_class_init):
          * gst-libs/gst/rtp/gstbasertppayload.c:
          (gst_basertppayload_class_init):
          * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
          * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
          * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
          * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
          * gst/playback/gststreamselector.c:
          (gst_stream_selector_class_init):
          * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
          * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
          * sys/v4l/gstv4lcolorbalance.c:
          (gst_v4l_color_balance_channel_class_init):
          * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
          * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
          * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
          * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
          (gst_v4l_tuner_norm_class_init):
          * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
          * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
          * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
          Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)

2006-04-08 18:09:17 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          Fix broken GObject macros
          Original commit message from CVS:
          * ext/pango/gsttextrender.h:
          * gst-libs/gst/audio/gstaudiosink.h:
          * gst-libs/gst/audio/gstaudiosrc.h:
          * gst-libs/gst/audio/gstbaseaudiosink.h:
          * gst-libs/gst/audio/gstbaseaudiosrc.h:
          * gst-libs/gst/audio/gstringbuffer.h:
          * gst-libs/gst/rtp/gstbasertpdepayload.h:
          * gst-libs/gst/rtp/gstbasertppayload.h:
          * gst-libs/gst/video/gstvideofilter.h:
          * gst-libs/gst/video/gstvideosink.h:
          * gst/playback/gstplaybasebin.h:
          * gst/tcp/gstmultifdsink.h:
          * sys/v4l/gstv4lelement.h:
          Fix broken GObject macros

2006-04-08 16:21:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
          Original commit message from CVS:
          * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
          More debug to trace why my USB headset is not working with gst

2006-04-07 17:18:11 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (group_destroy):
          Clean up our group elements properly in the case where it never
          got committed - it still got added unconditionally to the bin.

2006-04-07 15:14:32 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/theora/theoradec.c: Unref unhandled events.
          Original commit message from CVS:
          * ext/theora/theoradec.c: (theora_dec_sink_event),
          (theora_handle_data_packet), (theora_dec_chain):
          Unref unhandled events.
          Protect against empty buffers.
          Perform QoS on running time.

2006-04-07 13:24:54 +0000  Michael Smith <msmith@xiph.org>

          ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
          Original commit message from CVS:
          * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
          (gst_vorbis_enc_chain):
          Remove leaks from vorbisenc.
          Mostly minor changes, the only significant one is that now the
          buffers we set as 'streamheader' on the caps are copies of the
          original buffers, to avoid circular refcounting problems.

2006-04-07 09:51:35 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-04-06 11:40:45 +0000  Andy Wingo <wingo@pobox.com>

        * 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.

2006-04-04 11:20:24 +0000  Andy Wingo <wingo@pobox.com>

          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!

2006-04-04 11:15:00 +0000  Andy Wingo <wingo@pobox.com>

          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.

2006-04-04 10:16:46 +0000  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 ...
          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.

2006-04-03 16:43:10 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-04-03 12:55:18 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-04-03 08:49:06 +0000  Andy Wingo <wingo@pobox.com>

          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.

2006-04-03 08:32:21 +0000  Andy Wingo <wingo@pobox.com>

          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.

2006-04-03 08:28:58 +0000  Andy Wingo <wingo@pobox.com>

          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.

2006-04-01 15:34:38 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-04-01 11:41:13 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-04-01 11:21:30 +0000  Thomas Vander Stichele <thomas@apestaart.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)

2006-04-01 09:50:34 +0000  Thomas Vander Stichele <thomas@apestaart.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

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>

        * 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

2006-03-30 10:03:56 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-03-29 14:00:08 +0000  j^ <j@bootlab.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

2006-03-29 13:54:24 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-03-28 13:13:43 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
          Original commit message from CVS:
          * gst/playback/gstplaybin.c: (gen_video_element),
          (gen_text_element), (gen_audio_element), (gen_vis_element):
          Error out gracefully when we can't create any of the usual
          conversion elements for some reason. Also, don't try to
          create an audioscale (sic) element that's not used anyway.

2006-03-28 10:21:52 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (setup_source):
          Don't post RESOURCE_NOT_FOUND error when we can't find a source
          element for a particular protocol, that's confusing for users.
          Instead, post a RESOURCE_FAILED error, so that our own error
          message is actually shown in totem etc. (#336303).

2006-03-27 16:36:46 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

          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).

2006-03-27 16:15:00 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-03-24 20:35:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-03-24 17:57:39 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
          Original commit message from CVS:
          * gst/subparse/gstsubparse.c: (convert_encoding),
          (gst_sub_parse_change_state):
          * gst/subparse/gstsubparse.h:
          Text subtitle files may or may not be UTF-8. If it's not, we
          don't really want to see '?' characters in place of non-ASCII
          characters like accented characters. So let's assume the input
          is UTF-8 until we come across text that is clearly not. If it's
          not UTF-8, we don't really know what it is, so try the following:
          (a) see whether the GST_SUBTITLE_ENCODING environment variable
          is set; if not, check (b) if the current locale encoding is
          non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
          the current locale encoding is UTF-8 and the environment variable
          was not set to any particular encoding. Not perfect, but better
          than nothing (and better than before, I think) (fixes #172848).

2006-03-24 17:39:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * docs/plugins/tmpl/.gitignore:
        * tests/check/libs/.gitignore:
        * tests/check/pipelines/.gitignore:
        * tests/examples/volume/.gitignore:
          ignore more
          Original commit message from CVS:
          ignore more

2006-03-24 17:26:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
          Original commit message from CVS:
          2006-03-24  Thomas Vander Stichele <thomas at apestaart dot org>
          * configure.ac:
          update core requirement to 0.10.4.1 because of async_playback
          vmethod on GstBaseSink

2006-03-24 17:11:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          use DEBUG_FUNCPTR for collectpads
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
          * gst/adder/gstadder.c: (gst_adder_init):
          use DEBUG_FUNCPTR for collectpads

2006-03-24 14:11:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

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).

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).

2006-03-17 17:48:33 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-03-17 13:11:45 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-03-16 20:01:03 +0000  Michal Benes <michal.benes@xeris.cz>

          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)

2006-03-16 13:08:01 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
          Original commit message from CVS:
          * gst-libs/gst/tag/gstid3tag.c:
          Add TXXX frame identifiers for replaygain stuff as used
          by some taggers (see #323721).

2006-03-16 10:22:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-03-15 22:40:08 +0000  Tim-Philipp Müller <tim@centricular.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.

2006-03-15 17:59:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-03-15 17:11:34 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

          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).

2006-03-15 11:30:29 +0000  Sebastien Moutte <sebastien@moutte.net>

          gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
          Original commit message from CVS:
          Patch by: Sebastien Moutte  <sebastien moutte net>
          * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
          (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
          (gst_ffmpeg_smpfmt_to_caps):
          Replace __VA_ARGS__ caps creation macros with varargs functions.
          Makes things compile on MSVC (#320765), looks nicer, and we can
          tell the compiler to check for the NULL terminator.

2006-03-14 15:13:04 +0000  Fabrizio Gennari <fabrizio.ge@tiscali.it>

          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).

2006-03-14 13:16:49 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
          Add support for 24bpp DIB (#305279).

2006-03-14 11:11:59 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-03-14 09:51:01 +0000  Tim-Philipp Müller <tim@centricular.net>

          configure.ac: back to CVS.
          Original commit message from CVS:
          * configure.ac:
          back to CVS.

=== release 0.10.5 ===

2006-03-13 19:50:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2006-03-13 17:28:23 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-03-13 11:17:19 +0000  Tim-Philipp Müller <tim@centricular.net>

          docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
          Original commit message from CVS:
          * docs/plugins/Makefile.am:
          Part of previous cdparanoiasrc docs fixes, forgot to commit.

2006-03-12 14:56:31 +0000  Tim-Philipp Müller <tim@centricular.net>

          docs/plugins/: Add cdparanoiasrc to docs.
          Original commit message from CVS:
          * docs/plugins/gst-plugins-base-plugins-docs.sgml:
          * docs/plugins/gst-plugins-base-plugins-sections.txt:
          * docs/plugins/gst-plugins-base-plugins.hierarchy:
          Add cdparanoiasrc to docs.
          * gst-libs/gst/cdda/gstcddabasesrc.c:
          More GstCddaBaseSrc docs.

2006-03-12 13:47:22 +0000  Tim-Philipp Müller <tim@centricular.net>

          Add new API to libgsttag: gst_tag_from_id3_user_tag().
          Original commit message from CVS:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
          * gst-libs/gst/tag/tag.h:
          Add new API to libgsttag: gst_tag_from_id3_user_tag().

2006-03-11 19:47:16 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (plugin_init):
          NULL-terminate array of mpeg4 video file extensions.
          Fixes crash on PPC (#334226).

2006-03-11 16:40:20 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-03-10 19:15:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          configure.ac: back to trunk
          Original commit message from CVS:
          * configure.ac:
          back to trunk

=== release 0.10.4 ===

2006-03-10 19:05:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playbin.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * docs/upload.mak:
        * win32/common/config.h:
          releasing 0.10.4
          Original commit message from CVS:
          releasing 0.10.4

2006-03-10 12:37:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-03-10 11:09:23 +0000  Fabrizio <fabrizio.ge@tiscali.it>

          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)

2006-03-10 10:44:02 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-03-09 19:02:32 +0000  Thomas Vander Stichele <thomas@apestaart.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).

2006-03-09 15:05:03 +0000  Christophe Fergeau <teuf@gnome.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.

2006-03-09 12:56:35 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-03-09 12:37:59 +0000  Tim-Philipp Müller <tim@centricular.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).

2006-03-09 11:10:03 +0000  Tim-Philipp Müller <tim@centricular.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).

2006-03-08 17:11:29 +0000  Tim-Philipp Müller <tim@centricular.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).

2006-03-08 11:34:45 +0000  Michael Smith <msmith@xiph.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.

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.

2006-03-08 09:37:12 +0000  Fabrizio <fabrizio.ge@tiscali.it>

          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).

2006-03-07 15:08:15 +0000  Tim-Philipp Müller <tim@centricular.net>

          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().

2006-03-07 13:01:21 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
          Original commit message from CVS:
          * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
          Don't ignore flow return from gst_pad_push().

2006-03-07 12:49:03 +0000  Christophe Fergeau <teuf@gnome.org>

          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).

2006-03-06 20:52:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          ext/gnomevfs/gstgnomevfssink.c: change location param details
          Original commit message from CVS:
          * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
          change location param details
          * gst/volume/gstvolume.c: (plugin_init):
          correct plugin description

2006-03-06 20:07:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-03-06 16:18:51 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-03-06 14:14:47 +0000  Fabrizio <fabrizio.ge@tiscali.it>

          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).

2006-03-05 23:39:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          ext/ogg/gstoggmux.c: put Theora BOS pages before others.  This hardcodes the Ogg/Theora I profile, but hey.
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
          (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
          put Theora BOS pages before others.  This hardcodes
          the Ogg/Theora I profile, but hey.

2006-03-05 23:06:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ext/ogg/gstoggmux.c:
          changed more than 5 lines
          Original commit message from CVS:
          changed more than 5 lines

2006-03-05 22:57:58 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-03-05 21:34:23 +0000  Julien Moutte <julien@moutte.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)

2006-03-05 13:44:05 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-03-05 13:08:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
          Add support for '3IVD' fourcc (#333403).

2006-03-04 14:35:10 +0000  Tim-Philipp Müller <tim@centricular.net>

          configure.ac: Bump requirements to GStreamer CVS for the new error enum.
          Original commit message from CVS:
          * configure.ac:
          Bump requirements to GStreamer CVS for the new error enum.
          * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
          Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
          space left on the device (fixes #333352).

2006-03-03 23:53:50 +0000  Sébastien Moutte <sebastien@moutte.net>

          win32/vs6: add a project file for libgstvolume update the workspace
          Original commit message from CVS:
          * win32/vs6:
          add a project file for libgstvolume
          update the workspace

2006-03-03 15:26:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ext/ogg/README:
        * ext/ogg/gstoggmux.c:
          debug updates
          Original commit message from CVS:
          debug updates

2006-03-03 15:22:02 +0000  Thomas Vander Stichele <thomas@apestaart.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

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>

        * gst/videorate/Makefile.am:
          fix wim's commit
          Original commit message from CVS:
          fix wim's commit

2006-03-02 17:48:40 +0000  Thomas Vander Stichele <thomas@apestaart.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>

        * ext/ogg/gstoggmux.c:
          clean up debug messages
          Original commit message from CVS:
          clean up debug messages

2006-03-02 17:15:38 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2006-03-02 17:04:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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.

2006-03-02 12:35:59 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
          Original commit message from CVS:
          * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
          (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
          (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
          (gst_ogm_parse_chain):
          Don't remove/re-add source pad if the new caps are the same as
          the old caps anyway (#333042). When removing source pad, don't
          unref it afterwards - we didn't ref it when adding. Sprinkle some
          GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
          after using gst_pad_get_parent(). Return downstream flow return
          value in chain function.

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)

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.

2006-02-24 17:31:53 +0000  Michael Smith <msmith@xiph.org>

          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).

2006-02-24 16:21:34 +0000  Tim-Philipp Müller <tim@centricular.net>

          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>)

2006-02-24 13:54:04 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-24 12:18:14 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-24 10:18:52 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-02-24 00:07:18 +0000  Tim-Philipp Müller <tim@centricular.net>

        * ChangeLog:
          ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
          Original commit message from CVS:
          ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15

2006-02-22 18:46:46 +0000  Wim Taymans <wim.taymans@gmail.com>

          ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
          Original commit message from CVS:
          * ext/alsa/gstalsasrc.c:
          Fix description as pointed out by caugier.

2006-02-22 10:29:22 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-02-21 12:16:16 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-21 12:05:18 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-02-20 18:27:06 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
          Original commit message from CVS:
          * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
          (parse_subrip), (gst_sub_parse_format_autodetect):
          Set right caps given that we send escaped text. Also,
          honour <i></i>, <b></b> and <u></u> markers that can be found
          in .srt files (fixes #310202).

2006-02-20 16:21:14 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
          Original commit message from CVS:
          * gst-libs/gst/audio/mixerutils.c:
          (element_factory_rank_compare_func):
          Make order in which elements are tried more determinable.

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).

2006-02-19 14:09:40 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
          Original commit message from CVS:
          * gst/playback/gstdecodebin.c: (mimetype_is_raw):
          That should be text/x-pango-markup, not text/x-pango-layout.

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>

        * 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).

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).

2006-02-16 20:19:51 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
          Original commit message from CVS:
          * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
          Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
          with 4 or 6 channels, assume a default channel layout to make things
          work (not sure there's anything else we can do in those cases).

2006-02-16 19:18:46 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/multichannel.c: Minor docs fix.
          Original commit message from CVS:
          * gst-libs/gst/audio/multichannel.c:
          Minor docs fix.
          * gst-libs/gst/riff/Makefile.am:
          * gst-libs/gst/riff/riff-ids.h:
          * gst-libs/gst/riff/riff-media.c:
          (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
          Add support for WAVEFORMATEX, eg. PCM audio with more than two
          channels and a channel layout map.

2006-02-16 17:06:46 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-02-16 11:44:43 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-02-16 09:29:38 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-15 12:20:47 +0000  Andy Wingo <wingo@pobox.com>

          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.

2006-02-15 12:07:57 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-02-14 18:52:52 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-14 13:45:35 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
          Original commit message from CVS:
          * gst-libs/gst/audio/gstringbuffer.c:
          (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
          (gst_ring_buffer_samples_done), (wait_segment),
          (gst_ring_buffer_commit), (gst_ring_buffer_clear):
          Add some compiler G_(UN_)LIKELY help.
          SIGNAL the ringbuffer waiters when going to PAUSED as well to
          make sure they can exit their functions. Should fix #330748

2006-02-13 20:49:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-02-13 18:49:02 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-02-13 15:59:48 +0000  Wim Taymans <wim.taymans@gmail.com>

          gst/playback/gstplaybin.c: Update vis bin docs.
          Original commit message from CVS:
          * gst/playback/gstplaybin.c: (gen_vis_element):
          Update vis bin docs.
          Move queue after tee so we don't queue video buffers but
          audio samples instead. Fixes problems where the video queue
          is filled and the audio queue empty.

2006-02-13 15:17:34 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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

2006-02-09 12:16:35 +0000  Andy Wingo <wingo@pobox.com>

        * ChangeLog:
          changelog foo
          Original commit message from CVS:
          changelog foo

2006-02-09 11:46:03 +0000  Andy Wingo <wingo@pobox.com>

        * 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

=== release 0.10.3 ===

2006-02-09 11:18:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playbin.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
        * win32/common/config.h:
          Releasing 0.10.3
          Original commit message from CVS:
          Releasing 0.10.3

2006-02-08 18:37:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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)

2006-02-08 15:57:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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).

2006-02-08 15:50:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-02-08 12:46:14 +0000  Jan Schmidt <thaytan@mad.scientist.com>

        * 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

2006-02-08 11:04:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * gst/tcp/gstmultifdsink.c:
          doc fixes
          Original commit message from CVS:
          doc fixes

2006-02-08 09:20:23 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-02-07 18:32:00 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-07 17:44:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-07 16:16:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
          In case we can't find the required number of consecutive
          mpeg audio frames to positively identify an MPEG audio
          stream, check if there's at least a valid mpeg audio
          frame right at offset 0 and if so suggest mpeg/audio
          caps with a very low probability (#153004).

2006-02-07 15:52:26 +0000  Andy Wingo <wingo@pobox.com>

          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.

2006-02-07 14:57:46 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (prepare_output):
          Don't print the URI as part of the error message, it
          makes error dialogs look rather ugly, especially if
          the URI is very long or has characters in it that
          need escaping.

2006-02-07 13:11:31 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (prepare_output):
          Error out if we have only text or subtitles, but nothing
          else. Also error out if we have subtitles but no video
          stream.

2006-02-07 11:44:39 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
          Original commit message from CVS:
          * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
          Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
          Post an error message on the bus when we encounter an
          error, which will hopefully be more meaningful than the
          'Internal Flow Error' message users get to see if we
          just return GST_FLOW_ERROR.

2006-02-07 11:28:04 +0000  Andy Wingo <wingo@pobox.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).

2006-02-06 19:09:26 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-02-06 13:26:54 +0000  Julien Moutte <julien@moutte.net>

          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.

2006-02-06 12:38:48 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-02-05 23:31:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-02-05 22:47:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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"

2006-02-05 22:44:55 +0000  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.

2006-02-05 18:22:01 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-05 18:01:33 +0000  Tim-Philipp Müller <tim@centricular.net>

          Add libgsttag to the docs.
          Original commit message from CVS:
          * docs/libs/Makefile.am:
          * docs/libs/gst-plugins-base-libs-docs.sgml:
          * docs/libs/gst-plugins-base-libs-sections.txt:
          * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
          * gst-libs/gst/tag/gstvorbistag.c:
          * gst-libs/gst/tag/tag.h:
          * gst-libs/gst/tag/tags.c:
          Add libgsttag to the docs.

2006-02-05 17:21:23 +0000  Julien Moutte <julien@moutte.net>

          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.

2006-02-05 17:15:17 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2006-02-05 15:14:06 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-05 14:59:28 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-05 12:06:25 +0000  Julien Moutte <julien@moutte.net>

          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.

2006-02-05 03:05:41 +0000  David Schleef <ds@schleef.org>

          Add Dirac typefinding and add dirac format to oggmux.
          Original commit message from CVS:
          * ext/ogg/gstoggmux.c:
          * gst/typefind/gsttypefindfunctions.c:
          Add Dirac typefinding and add dirac format to oggmux.

2006-02-04 07:49:03 +0000  Michael Smith <msmith@xiph.org>

        * configure.ac:
          Improve error message for liboil missingness.
          Original commit message from CVS:
          Improve error message for liboil missingness.

2006-02-03 19:23:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-02-03 17:45:44 +0000  Edgard Lima <edgard.lima@indt.org.br>

        * 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.

2006-02-02 12:14:35 +0000  Wim Taymans <wim.taymans@gmail.com>

          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.

2006-02-02 08:53:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * tests/check/pipelines/theoraenc.c:
          comment on my understanding
          Original commit message from CVS:
          comment on my understanding

2006-02-02 08:47:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * common:
        * tests/check/pipelines/theoraenc.c:
          reformat to fit 80 chars
          Original commit message from CVS:
          reformat to fit 80 chars

2006-02-02 00:04:37 +0000  Kai Vehmanen <kv2004@eca.cx>

          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.

2006-02-01 14:51:29 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
          Original commit message from CVS:
          * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
          Make test work again by connecting fakesinks to each decoded pad,
          which makes the pipeline wait until each fakesink has a buffer
          queued before going to PAUSED state. At that point we know the
          decodebin pads are negotiated.

2006-02-01 11:59:47 +0000  Tim-Philipp Müller <tim@centricular.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.

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.

2006-02-01 09:58:15 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
          Original commit message from CVS:
          * gst/typefind/gsttypefindfunctions.c: (au_type_find),
          (paris_type_find), (ilbc_type_find), (plugin_init):
          Fix typefinding for audio/x-au, audio/x-paris and
          audio/iLBC-sh. We cannot use the START_WITH macros
          here, because there can only be one typefind factory
          with the same name (caps), so the second one would
          replace the first one and the first one would never
          be called when doing typefinding (see #161712).

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

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>

        * gst-plugins-base.spec.in:
          update with latest files
          Original commit message from CVS:
          update with latest files

2006-01-30 23:42:54 +0000  Sébastien Moutte <sebastien@moutte.net>

          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

2006-01-30 22:18:53 +0000  Sébastien Moutte <sebastien@moutte.net>

          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

2006-01-30 19:22:22 +0000  Andy Wingo <wingo@pobox.com>

          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.

2006-01-30 19:21:07 +0000  Andy Wingo <wingo@pobox.com>

        * 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

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)

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>

          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

2006-01-29 18:07:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * configure.ac:
          generate win32/common/config.h
          Original commit message from CVS:
          generate win32/common/config.h

2006-01-29 18:05:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-01-28 18:22:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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.

2006-01-28 18:19:18 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-01-28 16:35:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-01-28 02:13:14 +0000  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...
          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.

2006-01-27 23:40:19 +0000  David Schleef <ds@schleef.org>

          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

2006-01-27 17:00:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-01-27 01:36:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

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

2006-01-26 23:21:31 +0000  Julien Moutte <julien@moutte.net>

          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.

2006-01-26 20:40:20 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

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.

2006-01-21 22:40:03 +0000  Julien Moutte <julien@moutte.net>

          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.

2006-01-20 19:10:26 +0000  Christian Schaller <uraeus@gnome.org>

        * autogen.sh:
        * common:
          up automake requirement to 1.7
          Original commit message from CVS:
          up automake requirement to 1.7

2006-01-19 10:59:51 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst/playback/gstplaybasebin.c: Free iterator when done.
          Original commit message from CVS:
          * gst/playback/gstplaybasebin.c: (setup_source):
          Free iterator when done.

2006-01-17 11:43:49 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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

2006-01-16 21:01:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * configure.ac:
          back to HEAD
          Original commit message from CVS:
          back to HEAD

=== release 0.10.2 ===

2006-01-16 20:59:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * docs/plugins/gst-plugins-base-plugins.args:
        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-decodebin.xml:
        * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
        * docs/plugins/inspect/plugin-gnomevfs.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-playbin.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-video4linux.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
          releasing 0.10.2
          Original commit message from CVS:
          releasing 0.10.2

2006-01-16 16:38:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-01-16 15:31:14 +0000  Thomas Vander Stichele <thomas@apestaart.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)

2006-01-16 15:19:55 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-01-15 10:06:40 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-01-14 22:59:49 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-01-14 20:46:25 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-01-14 12:52:22 +0000  Tim-Philipp Müller <tim@centricular.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.

2006-01-13 17:17:07 +0000  Tim-Philipp Müller <tim@centricular.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).

2006-01-13 16:45:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

2006-01-13 16:17:50 +0000  Christian Schaller <uraeus@gnome.org>

        * gst-plugins-base.spec.in:
          remove version suffix
          Original commit message from CVS:
          remove version suffix

2006-01-12 14:56:11 +0000  Tim-Philipp Müller <tim@centricular.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.

2006-01-11 18:30:25 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          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.

2006-01-11 18:03:24 +0000  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 ...
          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.

2006-01-11 17:38:35 +0000  Christian Schaller <uraeus@gnome.org>

        * gst-plugins-base.spec.in:
          update spec.in file
          Original commit message from CVS:
          update spec.in file

2006-01-11 15:11:20 +0000  Tim-Philipp Müller <tim@centricular.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).

2006-01-10 23:58:36 +0000  Thomas Vander Stichele <thomas@apestaart.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

2006-01-10 18:06:56 +0000  Tim-Philipp Müller <tim@centricular.net>

          ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
          Original commit message from CVS:
          * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
          (gst_cd_paranoia_paranoia_callback):
          Don't try to free NULL pointers.

2006-01-10 15:47:48 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2006-01-10 12:25:59 +0000  Michael Smith <msmith@xiph.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.

2006-01-10 11:49:28 +0000  Jan Schmidt <thaytan@mad.scientist.com>

          gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
          Original commit message from CVS:
          * gst/playback/gstplaybin.c: (gen_audio_element):
          Don't leak an autoaudiosink/alsasink when we generate
          a new audio element. (old code, I guess)

2006-01-10 11:04:21 +0000  Michael Smith <msmith@xiph.org>

          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.

2006-01-10 10:06:53 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-01-10 09:38:44 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2006-01-09 10:52:33 +0000  Tim-Philipp Müller <tim@centricular.net>

          gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
          Original commit message from CVS:
          * gst-libs/gst/audio/Makefile.am:
          Link against libgstinterfaces, needed for mixer
          and property probe stuff.

2006-01-09 10:46:52 +0000  Edward Hervey <bilboed@bilboed.com>

          gst-libs/gst/Makefile.am:
          Original commit message from CVS:
          * gst-libs/gst/Makefile.am:

2006-01-09 09:38:34 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2006-01-03 17:33:16 +0000  Michael Smith <msmith@xiph.org>

          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

2006-01-03 17:27:13 +0000  Michael Smith <msmith@xiph.org>

          gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
          Original commit message from CVS:
          * gst/audioresample/resample.h:
          Declare struct _ResampleState.buffer as unsigned char *, not void *,
          since we do arithmetic on it.

2006-01-02 23:37:38 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2005-12-30 14:54:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

          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

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.

2005-12-29 14:31:49 +0000  Michael Smith <msmith@xiph.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.

2005-12-29 12:43:22 +0000  Tim-Philipp Müller <tim@centricular.net>

          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)

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).

2005-12-29 12:13:57 +0000  Tim-Philipp Müller <tim@centricular.net>

          tests/check/Makefile.am: Fix test case compilation.
          Original commit message from CVS:
          * tests/check/Makefile.am:
          Fix test case compilation.

2005-12-29 11:49:11 +0000  Tim-Philipp Müller <tim@centricular.net>

          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).

2005-12-28 18:19:25 +0000  Tim-Philipp Müller <tim@centricular.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.

2005-12-28 18:06:50 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2005-12-28 14:59:41 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2005-12-27 22:29:43 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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.

2005-12-23 18:16:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * configure.ac:
          back to HEAD
          Original commit message from CVS:
          back to HEAD

=== release 0.10.1 ===

2005-12-23 18:08:39 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2005-12-21 20:59:52 +0000  Edgard Lima <edgard.lima@indt.org.br>

        * ChangeLog:
        * gst/typefind/gsttypefindfunctions.c:
          iLBC30 and iLBC20 added to typefind.
          Original commit message from CVS:
          iLBC30 and iLBC20 added to typefind.

2005-12-20 15:57:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2005-12-20 12:24:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2005-12-20 12:00:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

          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

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

2005-12-18 10:54:33 +0000  Julien Moutte <julien@moutte.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.

2005-12-18 00:56:07 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

          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)

2005-12-18 00:41:10 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

          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)

2005-12-16 22:00:07 +0000  Stefan Kost <ensonic@users.sourceforge.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

2005-12-16 12:32:37 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2005-12-16 11:25:51 +0000  Benjamin Pineau <ben.pineau@gmail.com>

          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).

2005-12-15 14:43:38 +0000  Michael Smith <msmith@xiph.org>

          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.

2005-12-15 10:30:14 +0000  Michael Smith <msmith@xiph.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.

2005-12-15 10:15:10 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2005-12-15 09:48:19 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

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).

2005-12-12 20:31:24 +0000  Christian Schaller <uraeus@gnome.org>

        * ChangeLog:
        * ext/Makefile.am:
        * gst-plugins-base.spec.in:
          updates to activate cdparanoia plugin
          Original commit message from CVS:
          updates to activate cdparanoia plugin

2005-12-12 19:13:09 +0000  Michael Smith <msmith@xiph.org>

          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.

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.

2005-12-11 22:26:08 +0000  Julien Moutte <julien@moutte.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.

2005-12-09 10:23:42 +0000  Michael Smith <msmith@xiph.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.

2005-12-08 18:53:57 +0000  Luca Ognibene <luogni@tin.it>

          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.

2005-12-07 18:51:35 +0000  Tim-Philipp Müller <tim@centricular.net>

          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.

2005-12-07 11:34:37 +0000  Edward Hervey <bilboed@bilboed.com>

          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.

2005-12-06 19:42:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * 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

2005-12-06 19:28:24 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * docs/libs/tmpl/gstaudio.sgml:
        * docs/libs/tmpl/gstcolorbalance.sgml:
        * docs/libs/tmpl/gstgconf.sgml:
        * docs/libs/tmpl/gstmixer.sgml:
        * docs/libs/tmpl/gstringbuffer.sgml:
        * docs/libs/tmpl/gsttuner.sgml:
        * docs/libs/tmpl/gstxoverlay.sgml:
          put back stability level
          Original commit message from CVS:
          put back stability level

2005-12-05 18:11:49 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * configure.ac:
          back to HEAD
          Original commit message from CVS:
          back to HEAD

=== release 0.10.0 ===

2005-12-05 18:02:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * 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