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=== release 0.10.29 ===

2010-04-28  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  releasing 0.10.29, "Freaks"

2010-04-28 01:34:24 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-04-25 23:14:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
  0.10.28.3 pre-release

2010-04-20 17:20:43 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-plugins-base.doap:
  doap: update repository info from cvs->git and maintainers

2010-04-23 14:39:46 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
  Automatic update of common submodule
  From fc85867 to 4d67bd6

2010-04-22 20:58:29 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/ffmpegcolorspace/imgconvert.c:
  ffmpegcolorspace: Fix Y41B->Y444 conversion
  ...which is the intermediate conversion for conversion to all
  other formats.
  Fixes bug #616545.

2010-04-16 20:03:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
  audiorate: Don't leak the input buffer in error cases
  Fixes bug #615572.

2010-03-29 12:53:11 +0300  Stefan Kost <ensonic@users.sf.net>

* ext/ogg/gstoggmux.c:
  docs: fix typo in link name

2010-04-15 12:59:53 +0300  Stefan Kost <ensonic@users.sf.net>

* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
  x(v)imagesink: gracefully handle ximagesink>xwindow == NULL
  Expose could be called before we have set the xwindow. Handle this gracefully
  like we do in image_put.
  Fixes #615789

2010-04-15 11:44:49 +0300  Stefan Kost <ensonic@users.sf.net>

* sys/ximage/ximagesink.c:
  ximagesink: refactor _update_geometry()
  Refactor like in xvimagesink. Remove the extra parameter and adjust the assert check.

2010-04-15 07:18:05 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* configure.ac:
  configure: Drop -Wcast-align
  Commit message copied from core's commit from Benjamin Otte:
  246f5dba96a5b50bb74621af67b30942cca72af5
  Apparently gcc warns that GstMiniObject is not castable to
  GstEvent/Message/Buffer due to them containing 64bit variables, even
  though ARM hackers claim that those only need 4byte alignment. And as
  long as gcc behaves that way, this warning is not very useful.
  So we'll remove the warning until this problem is fixed.
  Fixes #615698

2010-04-14 14:13:25 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* gst-libs/gst/tag/lang-tables.dat:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/video-enumtypes.c:
  0.10.28.2 pre-release

2010-04-14 13:50:21 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-04-13 16:20:10 +0300  Stefan Kost <ensonic@users.sf.net>

* sys/xvimage/xvimagesink.c:
  xvimagesink: init geometry when setting new xid
  Don't rely on expose event to query geomentry after new xid is set.
  Fixes #615647.

2010-04-14 13:43:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/audioconvert/Makefile.am:
* tests/examples/app/Makefile.am:
* tests/examples/dynamic/Makefile.am:
* tests/examples/gio/Makefile.am:
* tests/examples/volume/Makefile.am:
* tests/old/examples/switch/Makefile.am:
  build: use LDADD instead of LDFLAGS to specify libs to link to when building executables
  Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
  This should make sure arguments are passed to the linker in the right
  order, and makes LDFLAGS usable again.
  Based on initial patch by Brian Cameron <brian.cameron@oracle.com>
  Fixes #615697.

2010-04-12 14:02:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefinding: add channels and rate to ADTS caps if we can

2010-04-12 13:33:18 +0100  Arun Raghavan <arun.raghavan@collabora.co.uk>

* gst/typefind/Makefile.am:
* gst/typefind/gstaacutil.c:
* gst/typefind/gstaacutil.h:
* gst/typefind/gsttypefindfunctions.c:
  typefinding: add AAC level to ADTS caps
  This adds code to calculate the level for a given AAC stream and export
  it in the stream caps. For AAC LC streams, the level is calculated
  according to the definition under the AAC Profile. For other streams,
  the definition under the Main Profile is used.
  HE-AAC support is still to be done, and is dependent on detecting the
  presence of SBR and PS in the stream.
  Level is added as a field of type string because that's the way it's
  done in H.264 caps as well. There are only a few possible levels, so
  not using a numerical type is not too painful in this case, and
  consistency is nice.
  Fixes #613589.

2010-03-10 13:32:53 +0000  Arun Raghavan <arun.raghavan@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefinding: add AAC profile to ADTS caps
  This looks at the AAC profile for ADTS streams and adds the profile as a
  string in the corresponding caps.
  Profile is the actual profile, base-profile denotes the minimum codec
  requirements to decode this stream. In this case they're always the
  same, but they may differ e.g. in case of certain HE-AAC streams that
  can be partially decoded by LC decoders (with loss of quality of course)
  if no suitable HE-AAC decoder is available.
  Fixes #612312.

2010-04-11 22:58:15 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/adder/gstadder.c:
  adder: add support for negative playback rates
  Decrement sample counter when playing backwards. Set proper segment when playing
  backwards (0..cur instead or cur..-1). Add more logging and fix a format string.

2010-03-26 19:00:47 -0400  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiopayload: use ptime-multiple
  Based on patch by Olivier Crête <olivier.crete@collabora.co.uk>
  Fixes #613248

2010-04-09 16:06:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
  audiopayload: add property to control packet duration
  Add a property to specify that the amount of data in a packet should be a
  multiple of ptime-multiple.
  See #613248

2010-04-09 11:20:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 218568f to fc85867

2010-04-08 17:49:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/Makefile.am:
* gst/playback/Makefile.am:
* gst/playback/gstplayback.h:
  playback, ogg: dist new gstplayback.h and gstogg.h

2010-04-09 08:23:33 +0200  Thomas Green <thomasgr33n@googlemail.com>

* gst/playback/gstplaybin.c:
  playbin: Only unref the volume element on dispose and when a new audio sink is set
  Unreffing it whenever the sinks are removed will make the volume
  element unavailable after a playbin reuse because it is only
  recreated if the audio sink has changed.
  Fixes bug #614288.

2010-04-08 07:39:08 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: Be sure that metadata is writable before setting caps
  Call gst_buffer_make_metadata_writable before attempting
  to set caps on the buffer.

2010-04-08 12:21:50 +0200  Edward Hervey <bilboed@bilboed.com>

* ext/gio/gstgio.c:
* ext/gnomevfs/gstgnomevfs.c:
  ext: Invert rank of gio and gnomevfs elements

2010-04-08 01:26:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
  alsa: don't pass non-constant strings as printf format strings
  Fixes 'format not a string literal and no format arguments' compiler
  warning when compiling with -DGST_DISABLE_PRINTF_EXTENSION.

2010-04-07 20:21:14 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/video/video.h:
  docs: add gtk-doc chunks with Since: tags for new GST_VIDEO_CAPS_GRAY* API

2010-04-07 19:07:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* autogen.sh:
* configure.ac:
  build: bump autoconf requirement to 2.60 for gobject-introspection.m4
  Require autoconf 2.60 (which was released in June 2006).
  Fixes #600718.

2010-04-07 17:25:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/video/video.c:
  video: Fix parsing of 8-bit grayscale caps

2010-04-07 17:21:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.h:
  video: API: Add GST_VIDEO_CAPS_GRAY{8,16}

2010-04-07 17:08:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-sections.txt:
  video: API: Add gst_video_format_is_gray() to the docs

2010-04-07 17:07:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* win32/common/libgstvideo.def:
  video: Add new symbol to the exported symbols list

2010-04-07 17:06:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
  video: Add support for 8-bit and 16-bit grayscale formats

2010-04-06 10:55:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
  rtspconnection: Handle closed POST socket in tunneling
  Catch more socket errors.
  Rework how sockets are managed in the GSource, wake up the maincontext instead
  of adding/removing the sockets from the source.
  Add callback for when the tunnel connection is lost. Some clients (Quicktime
  Player) close the POST connection in tunneled mode and reopen the socket when
  needed.
  See #612915

2010-04-04 21:24:44 -0700  David Schleef <ds@schleef.org>

* configure.ac:
  configure: fix cdparanoia check
  Linking with libcdda_paranoia.so requires also linking with
  libcdda_interface.so.

2010-04-04 18:00:23 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* tests/check/libs/tag.c:
  tests: tag: Refactor a bit
  Refactor xmp tags unit tests and remove an useless assertion.
  This will make easier to add unit tests to serialize/deserialize
  taglists.

2010-04-04 21:18:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
  alsa: Ignore errors when unpreparing or closing the device
  Errors could happen here when the device was removed already
  or when something is broken anyway. If errors happen here and
  they're propagated, the element can't shutdown cleanly.
  Fixes bug #614545.

2010-04-04 20:55:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/alsa/gstalsamixer.c:
  alsamixer: Detect errors from device polling, stop the task and post an error message
  Partially fixes bug #614545.

2010-04-04 12:13:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* tests/examples/seek/Makefile.am:
  examples: build silly joystick seek example only on linux
  jsseek depends on linux headers and should therefore only be built
  on linux.
  Fixes #614764.

2010-04-03 22:49:11 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/audiotestsrc/gstaudiotestsrc.c:
  audiotestsrc: swap timestamps in forward and reverse mode.
  In reverse mode we want use the next next timestamp (and not the other way
  around). Fixes the tests again. Also readd a log line that was dropped with
  previous commit.

2010-04-03 14:03:45 +0100  Vincent Untz <vuntz@gnome.org>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  libs: point gobject-introspection scanner to .la files
  Point g-ir-scanner to the .la file of our library, which hopefully
  makes it find the right dependencies in all cases (ie. our locally
  built libgstreamer and not the system-installed one). This is also
  how it's done in Gtk+ and how it's documented in the wiki, see
  http://live.gnome.org/GObjectIntrospection/AutotoolsIntegration
  Fixes #603710.

2010-04-02 21:01:25 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
  audiotestsrc: implement reverse playback
  Support playback at negative rates. When having a GstController assigned, the
  element will produce time dependend output.

2010-04-02 20:56:19 +0300  Stefan Kost <ensonic@users.sf.net>

* tests/icles/audio-trickplay.c:
  tests: extend audio-trickplay test app
  Tell status in top comment. Use debug logging instead of print to be able to
  see timing issue in debug log viewer. Add more commandline flags. Test reverse
  playback.

2010-04-02 18:56:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/examples/seek/seek.c:
  seek: Only use embed_xid if HAVE_X is defined
  Fixes bug #614622.

2010-04-01 19:13:22 +0200  Edward Hervey <bilboed@bilboed.com>

* tests/check/pipelines/basetime.c:
  tests/basetime: Don't run test with osxaudiosrc
  libcheck runs the actual tests in a forked process and that makes the guys
  in Cupertino really sad.

2010-04-01 18:51:17 +0200  Edward Hervey <bilboed@bilboed.com>

* tests/check/pipelines/capsfilter-renegotiation.c:
  tests: Unref the bus once we're done with it

2010-04-01 16:49:37 +0200  Edward Hervey <bilboed@bilboed.com>

* common:
  common: Update for new suppressions

2010-04-01 13:55:15 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstplaysink.c:
  gstplaysink: Remove unused variable.
  The value of klass is never used

2010-04-01 13:53:37 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstdecodebin2.c:
  decodebin2: Removing dead assignment.
  The value of group is overwritten a few lines below before being used.

2010-04-01 13:51:13 +0200  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/tag/gsttagdemux.c:
  tagdemux: Remove unused variable

2010-04-01 13:48:42 +0200  Edward Hervey <bilboed@bilboed.com>

* ext/gnomevfs/gstgnomevfssink.c:
  gstgnomevfssink: Return the proper GstFlowReturn.
  We were always returning GST_FLOW_OK previously even if we encountered errors.

2010-03-30 23:44:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  gst-libs: more gobject-introspection fixes
  Use right .pc file variable for compiler includes this time:
  g-ir-compiler wants the girdirs not the typelibdirs as includes.

2010-03-30 20:21:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/seek/jsseek.c:
  examples: fix printf format warning in jsseek example
  Yes, I know about G_GSIZE_FORMAT.

2010-03-30 19:56:56 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  gst-libs: fix up gobject-introspection some more
  Use new girdir and typlibdir from core .pc files, so we can figure
  out the right includes to pass to the gobject-introspection tools,
  whether core is installed in the same prefix as gobject-introspection
  or in a different prefix or uninstalled. This also keeps us from adding
  bogus paths to the includes that only work if core is uninstalled.
  Also add some missing includes/pkgs where needed.

2010-03-30 19:29:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/Makefile.am:
  Our RIFF library depends on both the audio and tag libraries
  Update rules in Makefile.am accordingly.

2010-03-30 15:10:42 +0200  Robert Swain <robert.swain@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Fix aduio_raw_sink typo

2009-11-28 21:03:44 +0100  Jan Schmidt <thaytan@noraisin.net>

* tests/examples/seek/.gitignore:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/jsseek.c:
  examples: Add a silly joystick based shuttle example

2010-03-29 20:07:52 -0700  David Schleef <ds@schleef.org>

* ext/theora/gsttheoraenc.c:
  theoraenc: 0-length packets are delta units

2010-03-29 10:47:31 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/Makefile.am:
  gst-libs: build independent sub-directories in parallel if make -jN is used
  Build those libraries that don't depend on any other gst-plugins-base
  libraries in parallel if make -jN is used.

2010-03-29 00:22:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
* ext/Makefile.am:
* gst/Makefile.am:
* sys/Makefile.am:
* tests/examples/Makefile.am:
  build: build plugin and example directories in parallel if make -jN is used
  We know our plugins and examples are independent of each other, so may
  just as well build them in parallel. Makes the output a bit messy, but
  that shouldn't be a problem and can easily be avoided with make -j1.

2010-03-28 21:50:58 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/Makefile.am:
  gst-libs: specify dependencies in Makefile.am to make them explicit

2010-03-24 09:59:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
* sys/xvimage/xvimagesink.c:
* tests/icles/test-xoverlay.c:
  xoverlay: change new set_render_rectangle() vfunc to take four arguments so we don't depend on libgstvideo
  Don't make libgstinterfaces (and thus libgstaudio etc.) indirectly depend
  on libgstvideo by using the GstVideoRectangle helper structure in the API,
  which causes undesirable dependencies, esp. with the gobject-introspection
  (people will point and laugh at us if they find out that libgstaudio
  depends on libgstvideo). Instead, pass the x, y, width and height parameters
  directly to the function.
  Re-fixes #610249.

2010-03-25 18:45:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: we can handle avi in download mode too
  Add avi to the whitelisted types that can be used for download buffering.

2010-03-26 15:57:39 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  oggdemux: Provide packet duration function for old FLAC mapping too
  Fixes bug #613809.

2010-03-18 22:12:40 +0000  Damien Lespiau <damien.lespiau@intel.com>

* autogen.sh:
  autogen.sh: Don't call configure with --enable-plugin-docs
  configure gives a nice warning:
  configure: WARNING: unrecognized options: --enable-plugin-docs
  and indeed, I could not find anything in the configure.ac or the m4
  macros that would allow enabling that option. Remove it then.

2010-03-24 23:04:43 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
  tag: xmp: Do not remove tag from list twice
  There was a but when parsing the tags that removed two tags
  from the list when only one was parsed

2010-03-24 14:43:21 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
  tag: xmp: Add some comments
  Just adds some comments explaining some stuff about the
  (de)serialization functions. Add myself to the copyright list too.

2010-03-24 10:18:13 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
  tag: xmp: Adds _USER_RATING mapping for xmp
  Adds a new mapping for _USER_RATING on xmp helper lib
  and also adds tests for it

2010-03-23 09:32:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
  tag: xmp: Add Elevation tag mapping
  Adds a mapping to the _ELEVATION tag, this is a different
  mapping as it has to be mapped into exif:GPSAltitude and
  exif:GPSAltitudeRef at the same time. So we needed to refactor
  a little more to be able to deserialize it properly.
  Now, when parsing a xmp buffer into a taglist all tags are
  added to a list before being parsed so that when one of the
  altitude tags are found the deserialization function can search
  for its complementary tag to do the correct parsing
  Fixes #613690

2010-03-23 09:48:19 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
  tag: xmp: Fix off by one
  Avoid ignoring single char tags, like exif:GPSAltitudeRef
  Fixes #613690

2010-03-22 15:18:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
* tests/check/libs/tag.c:
  tag: xmp: Adds mappings for LATITUDE and LONGITUDE
  Adds the mappings for those tags and tests
  for tags serialization.
  Fixes #613690

2010-03-22 22:03:09 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
  tag: xmp: Refactor buffer parsing
  When parsing the xmp buffer into the gst taglist store the
  found tags into a list to be parsed only after finding all
  tags on the buffer. This allows the parser function to search
  this list for complimentary tags that should be parsed together
  Fixes #613690

2010-03-20 11:17:38 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
  tag: xmp: Refactor mappings storage
  This commit is only refactoring, no fetaures added.
  Do not store tags in flexible arrays as it doesn't allow us
  to use nested flexible arrays. This is going to be needed in the
  following commits to map gst tags that are stored into
  2 separate tags in xmp (Not that they are alternatives, but
  they are complementary).
  For example, GST_TAG_ELEVATION is represented in the exif
  schema with 2 fields: the absolute altitude and an integer
  to indicate if it is above or below sea level.
  The previous mappings storage wouldn't allow us to
  express it.
  Also store a serialization and a deserialization function
  for each xmp tag as some of them require some non-trivial
  convertion to its string form.
  Fixes #613690

2010-03-24 18:51:42 +0100  Edward Hervey <bilboed@bilboed.com>

* common:
  Automatic update of common submodule
  From 55cd514 to c1d07dd

2010-03-24 18:55:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* tests/examples/seek/seek.c:
  seek: parse more info from the buffering query
  Parse more info from the buffering query and log this as debug info.

2010-03-24 12:10:38 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtsptransport.c:
  rtsptransport: ignore unparsable ranges
  Ignore unparsable port ranges instead of erroring out.
  Fixes #613591

2010-03-23 18:36:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* win32/common/libgstrtsp.def:
  win32: Add new gst_rtsp_lower_trans_get_type() symbol to the symbol lists

2010-03-23 11:01:17 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst-libs/gst/riff/riff-media.c:
  riff: add some more fourcc for MPEG-4 video

2010-03-22 09:15:28 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  configure: require core git

2010-03-22 08:38:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-fft.pc.in:
  pkgconfig: Add @LIBM@ to the FFT pkg-config files

2010-03-22 08:35:57 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* pkgconfig/gstreamer-app-uninstalled.pc.in:
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
* pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
* pkgconfig/gstreamer-floatcast.pc.in:
* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
* pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
* pkgconfig/gstreamer-video-uninstalled.pc.in:
  pkgconfig: Fix include and library paths for the uninstalled pc files

2010-03-20 13:42:32 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/gio/gstgiobasesrc.c:
  gio: add cast to avoid compiler warning with old GLib versions
  g_file_input_stream_query_info() had char * instead of const char *
  as attribute argument before 2.20.
  Fixes #613387, spotted by tetsuyayasuda@gmail.com

2010-03-20 12:55:36 +0000  Torsten Schönfeld <kaffeetisch@gmx.de>

* gst-libs/gst/interfaces/xoverlay.c:
  docs: add Since: tags to gst_x_overlay_handle_event() docs
  Fixes #613403.

2010-03-19 22:33:58 +0100  Benjamin Otte <otte@redhat.com>

* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
  Constify some strings in the API
  Needed by plugins-good

2010-03-19 16:41:54 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: Only set color-matrix and chroma-site for relevant formats
  The color-matrix only makes sense for colorful formats, i.e. not Y800
  and the chroma-site only for non-4:4:4(:4) formats.

2010-03-19 15:37:04 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoradec.h:
  theoradec: add QoS messages to the decoder
  Post QoS messages when we drop a frame because of QoS.

2010-03-19 15:00:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtsptransport.h:
  rtsp: add GType for transport flags
  Make a method to register the transport flags as a GType.

2010-03-19 01:00:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/cdparanoia/Makefile.am:
* ext/gio/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
* gst/tcp/Makefile.am:
* gst/videotestsrc/Makefile.am:
* sys/v4l/Makefile.am:
* tests/examples/app/Makefile.am:
* tests/examples/overlay/Makefile.am:
* tests/icles/Makefile.am:
  build: Makefile.am fixes
  Mostly just add missing $(GST_BASE_CFLAGS), but also fix up order
  of flags (see docs/random/moving-plugins).

2010-03-19 00:46:56 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/check/pipelines/.gitignore:
  .gitignore: ignore new unit test binary

2010-03-17 23:57:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  configure.ac: -Wmissing-prototypes and -Wnested-externs are not valid for C++
  Fixes building Qt-based overlay examples in combination with -Werror.

2010-03-17 16:32:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  configure.ac: wrap overly long warning flag lines

2010-03-17 19:24:27 -0300  Reuben Dowle <reube.dowle@navico.com>

* sys/ximage/ximagesink.c:
  ximagesink: Fix caps leak
  Unref caps when peer doesn't accept caps
  Fixes #613198

2010-03-17 08:13:59 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* tests/check/Makefile.am:
* tests/check/pipelines/capsfilter-renegotiation.c:
  tests: capsfilter-renegotiation: Adds a new unit test
  Adds a new test for checking that capsfilter 'caps' property
  changes cause caps renegotiation on the pipeline.

2010-03-17 16:46:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_scanline.c:
  videoscale: Use correct boundary checks for YUY2/UYVY
  Fixes bug #613093.

2010-03-17 16:39:13 +0100  Peter Kjellerstedt <peter.kjellerstedt@axis.com>

* gst-libs/gst/rtsp/gstrtspdefs.c:
  rtsp: Further clean up of gst_rtsp_strresult()
  Since we no longer use an array of error messages, there is no reason
  to clamp the error code, which allows us to simplify the code some more
  and also to actually report the correct error code for unknown errors.

2010-03-17 15:41:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/volume/gstvolume.c:
  volume: Remove useless cast
  It's not necessary anymore after latest core change to GstValueArray.

2010-03-17 12:08:30 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
  Add more warning flags
  The warnings are:
  -Wcast-align
  -Winit-self
  -Wmissing-include-dirs
  -Waddress
  -Waggregate-return
  -Wno-multichar
  -Wnested-externs
  No code needed to be fixed.

2010-03-17 11:14:29 +0100  Benjamin Otte <otte@redhat.com>

* gst/audioconvert/gstfastrandom.h:
  Fix for -Wold-style-definition
  I didn't add the flag to configure because libvisual ships headers that
  trigger this warning.

2010-03-17 10:53:21 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
* ext/pango/gstclockoverlay.h:
* gst/subparse/mpl2parse.c:
  Add -Wformat-nonliteral -Wformat-security
  And fix the resulting compile failures.
  I'm sorry about the patch necessary to gstclockoverlay.h but after
  talking to Tim we decided we can live with it.

2010-03-17 10:51:57 +0100  Benjamin Otte <otte@redhat.com>

* gst-libs/gst/rtsp/gstrtspdefs.c:
  rtsp: Refactor gst_rtsp_strresult
  2 goals in the refactoring:
  - Put the error messages closer to their enum values, so that it's easy
  to see which error belongs to which value.
  - Make gcc not complain with -Wformat-nonliteral

2010-03-17 10:47:07 +0100  Benjamin Otte <otte@redhat.com>

* gst-libs/gst/tag/gstxmptag.c:
  xmp: Refactor code
  I initially looked here because I wanted compiles to not fail with
  -Wformat-nonliteral but ended up refactoring the code to make it look
  nicer.
  As I lack a large collection of XMP tagged files, I only did rough
  testing of the code. The testsuite passes though.

2010-03-16 20:05:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* m4/Makefile.am:
* m4/a52.m4:
* m4/aalib.m4:
* m4/as-arts.m4:
* m4/as-ffmpeg.m4:
* m4/as-liblame.m4:
* m4/as-slurp-ffmpeg.m4:
* m4/esd.m4:
* m4/gconf-2.m4:
* m4/glib.m4:
* m4/gst-artsc.m4:
* m4/gst-matroska.m4:
* m4/gst-sdl.m4:
* m4/gst-shout2.m4:
* m4/gst-sid.m4:
* m4/gtk.m4:
* m4/libfame.m4:
* m4/libmikmod.m4:
  m4: remove some unused .m4 files

2010-03-16 18:31:15 +0100  Benjamin Otte <otte@redhat.com>

* ext/alsa/gstalsaplugin.c:
* ext/ogg/gstoggdemux.c:
  More ENABLE_NLS fixes

2010-03-16 18:06:16 +0100  Benjamin Otte <otte@redhat.com>

* gst-libs/gst/gettext.h:
  Fix for ENABLE_NLS being undefined for -Wundef

2010-03-15 22:49:53 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
* ext/libvisual/visual.c:
* ext/theora/gsttheoraenc.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
* gst-libs/gst/tag/tags.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* gst/typefind/gsttypefindfunctions.c:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lelement.c:
* sys/xvimage/xvimagesink.c:
* tests/check/elements/audioconvert.c:
* tests/check/elements/gdpdepay.c:
* tests/check/elements/playbin.c:
* tests/check/elements/playbin2.c:
* tests/check/elements/videorate.c:
* tests/check/libs/pbutils.c:
* tests/check/libs/video.c:
* tests/check/pipelines/simple-launch-lines.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
* tests/icles/stress-playbin.c:
  Add -Wwrite-strings to configure
  Fixes for the code included

2010-03-16 15:45:23 +0100  Benjamin Otte <otte@redhat.com>

* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraparse.c:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdeclib.h:
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisparse.c:
* ext/vorbis/gstvorbistag.c:
* gst-libs/gst/sdp/gstsdpmessage.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstinputselector.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gststreamselector.c:
* gst/playback/gsturidecodebin.c:
* gst/subparse/gstssaparse.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/elements/audioconvert.c:
* tests/check/elements/playbin.c:
* tests/check/elements/playbin2.c:
* tests/check/elements/textoverlay.c:
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/pbutils.c:
* tests/old/testsuite/alsa/formats.c:
* tests/old/testsuite/alsa/sinesrc.c:
  gst_element_class_set_details => gst_element_class_set_details_simple
  Also change my email from the old university one to the current one.

2010-03-15 22:17:56 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
  Add -Wundef flag

2010-03-16 16:15:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtspconnection: allow for more ipv6 addresses
  Use hints in getaddrinfo() so that we can also resolve ipv6 addresses.

2010-03-11 14:52:09 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  baseaudiosink: arrange for a running ringbuffer/clock for _wait_eos
  Fixes #612223.

2010-03-16 01:08:48 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/check/elements/videorate.c:
  tests: fix videorate test
  Fix up videorate test for latest videotestsrc changes: just check for
  the important bits in the negotiated caps, not for exact equality with
  our filter caps. Also don't leak the videorate element in the test.

2010-03-15 12:54:32 -0500  Rob Clark <rob@ti.com>

* gst-libs/gst/riff/riff-media.c:
  riff: add mapping for On2 VP7 fourccs
  Fixes #612968.

2010-03-15 12:54:01 -0500  Rob Clark <rob@ti.com>

* gst-libs/gst/riff/riff-media.c:
  riff: add mapping for On2 VP62 fourcc
  See #612968.

2010-03-15 23:46:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/multichannel.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtsp/gstrtsptransport.h:
  docs: more helper libraries docs fixes
  Quieten gtk-doc a bit more.

2010-03-15 23:47:23 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspextension.c:
  docs: add GstRTSPExtension to docs
  Add minimal docs for GstRTSPExtension so people know it exists.

2010-03-15 18:45:13 +0000  David Hoyt <dhoyt@llnl.gov>

* gst/typefind/gsttypefindfunctions.c:
  typefind: use g_ascii_strncasecmp() instead of strncasecmp()
  g_ascii_strncasecmp() is more portable and likely more robust as
  well (with random binary data as input).
  Fixes #612845.

2010-03-15 13:39:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
  docs: fix typo in gst_tag_list_from_xmp_buffer() docs chunk

2010-03-15 13:32:58 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/interfaces/navigation.c:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
  docs: fix up interfaces library docs to make gtk-doc happy

2010-03-15 13:24:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-sections.txt:
  docs: add new libgstvideo API to documentation

2010-03-15 13:19:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* win32/common/libgstinterfaces.def:
* win32/common/libgstvideo.def:
  win32: add recently added API to .def files
  Also add API markers to make life easier for the release manager:
  API: gst_x_overlay_set_render_rectangle()
  API: gst_video_parse_caps_color_matrix()
  API: gst_video_parse_caps_chroma_site()

2010-03-15 13:14:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
  videotestsrc: use C comments instead of C++-style comments

2010-03-15 13:10:23 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: use g_value_set_static_string() for string constants

2010-03-15 14:26:28 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Avoid g_object_set() on NULL if a text sink is used
  Fixes bug #611702.

2010-03-15 14:10:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/subparse/gstsubparse.c:
  subparse: Correctly escape brackets in DKS regex
  Fixes bug #612783.

2010-03-15 11:36:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: make timeout usec more accurate
  Adjust the returned usec from the elapsed time so it represents the remaining
  timeout.

2010-03-15 11:41:35 +0200  Stefan Kost <ensonic@users.sf.net>

* tests/check/elements/videorate.c:
  tests: update videorate test for videotestsrc changes
  Add color-matrix to the caps we are comparing. Add logging og the caps in the
  test.

2010-03-15 01:35:15 -0700  David Schleef <ds@schleef.org>

* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
  videotestsrc: add chroma-zone-plate pattern
  pattern=chroma-zone-plate is pattern similar to zone-plate,
  but in the chroma channels instead of luma.

2010-03-15 01:34:09 -0700  David Schleef <ds@schleef.org>

* ext/theora/gsttheoradec.c:
  theoradec: add chroma-site to caps

2010-03-15 01:33:36 -0700  David Schleef <ds@schleef.org>

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: add chroma-site to caps

2010-03-15 01:31:20 -0700  David Schleef <ds@schleef.org>

* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
  video: add gst_video_parse_caps_chroma_site()

2010-03-14 19:10:16 -0700  David Schleef <ds@schleef.org>

* ext/theora/gsttheoradec.c:
  theoradec: add color-matrix to caps

2010-03-14 16:17:46 -0700  David Schleef <ds@schleef.org>

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: Add color-matrix to template caps

2010-03-14 22:14:19 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/seek/seek.c:
* tests/icles/test-colorkey.c:
* tests/icles/test-xoverlay.c:
  tests: make Gtk+ test programs compile with -DGSEAL_ENABLE
  Fixes #612552, at least for now.

2010-03-14 22:13:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* Makefile.am:
  build: add cruft alert for common/shave* leftovers to top-level Makefile.am

2010-03-14 13:11:53 -0700  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
  oggdemux: Don't drop zero-sized packets
  Zero-sized packets have relevence to Theora.

2010-03-12 15:47:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/volume/gstvolume.c:
  volume: Revert rounding behaviour changes when using controlled volume properties
  Now the controlled and non-controlled code paths are all having
  exactly the same rounding behaviour and the unit tests pass again.

2010-03-12 15:44:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/volume/gstvolume.c:
  volume: Only allocate a mute value array if a control source exists for the mute property

2010-03-12 13:55:55 +0100  Edward Hervey <bilboed@bilboed.com>

* common:
  Automatic update of common submodule
  From e272f71 to 55cd514

2010-03-10 10:50:32 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst-libs/gst/tag/gstxmptag.c:
  tags: Add new mapping to XMP helpers
  Adds geotagging mappings to XMP helpers
  Fixes #609539

2010-03-11 20:16:44 +0100  Benjamin Otte <otte@redhat.com>

* gst-libs/gst/interfaces/Makefile.am:
  Don't have 2 include dirs
  Seems to have been accidentally introduced in
  7269bc26d0a4bf44bd77a039fb54777625ef5f39.

2010-03-11 16:35:10 +0100  Edward Hervey <bilboed@bilboed.com>

* tests/icles/audio-trickplay.c:
  tests: Fix another unitialized variable

2010-03-11 16:09:26 +0100  Edward Hervey <bilboed@bilboed.com>

* tests/icles/audio-trickplay.c:
  tests: Fix unitialized variable.

2010-03-11 15:38:18 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
* ext/ogg/gstoggdemux.c:
* ext/theora/gsttheoraparse.c:
* ext/vorbis/gstvorbistag.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioquantize.h:
* gst/audioconvert/gstchannelmix.h:
* gst/playback/gstplaysink.c:
  Add -Wredundant-decls to warning flags
  ... and fix all the warnings that flag throws.

2010-03-11 13:32:14 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c:
* ext/ogg/gstogg.h:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstogmparse.c:
* ext/ogg/vorbis_parse.c:
* ext/ogg/vorbis_parse.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/riff/riff.c:
* gst-libs/gst/rtsp/gstrtspbase64.c:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/tag/lang.c:
* gst/ffmpegcolorspace/Makefile.am:
* gst/ffmpegcolorspace/gstffmpeg.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplayback.c:
* gst/playback/gstplayback.h:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/videorate/gstvideorate.h:
* tests/check/elements/appsink.c:
* tests/check/elements/audiorate.c:
* tests/check/elements/audioresample.c:
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/mixer.c:
* tests/check/libs/navigation.c:
* tests/examples/gio/giosrc-mounting.c:
  Add -Wmissing-declarations -Wmissing-prototypes to warning flags
  Includes all the fixes necessary to make stuff compile again.

2010-03-11 12:49:02 +0100  Benjamin Otte <otte@redhat.com>

* ext/gio/gstgiobasesink.c:
  gio: Remove unused function

2010-03-11 11:14:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/vorbis/gstvorbisparse.c:
  vorbisparse: make sure header buffer metadata is writable before modifying it
  Fixes unit test failures with core git.

2010-03-11 12:18:00 +0100  Benjamin Otte <otte@redhat.com>

* tests/check/elements/multifdsink.c:
  check: Ref buffers after setting caps on them
  Reffing makes metadata unwritable, so we need to set the caps before.

2010-03-11 12:04:32 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
  Add WARNING_CXXFLAGS where ERROR_CXXFLAGS are
  This matches the previous commit doing the same for CFLAGS in response
  to the common/ module changes.

2010-03-11 12:04:37 +0100  Edward Hervey <bilboed@bilboed.com>

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

2010-03-11 10:38:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/icles/test-xoverlay.c:
  tests: don't use Gtk+ 2.18 API for no good reason
  The rest of the code directly uses widget->allocation as well, so no point
  in using the new API in other places.

2010-03-11 11:20:48 +0100  Benjamin Otte <otte@redhat.com>

* common:
  Automatic update of common submodule
  From df8a7c8 to e272f71

2010-03-11 10:55:21 +0200  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/interfaces/xoverlay.c:
  xvoverlay: correct version number in docs

2010-02-26 13:56:21 +0200  Stefan Kost <ensonic@users.sf.net>

* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/audio-trickplay.c:
  tests: add a test for trickplay in audio synthesis graphs
  Right now this mostly demonstatest what not works. That is seeking with
  start-type = NONE to only update the rate and playing backwards. Also
  it shows that non-flushing seeks tend to lockup adder. Separate unit tests
  for the issues follow.

2010-02-08 17:20:35 +0200  Stefan Kost <ensonic@users.sf.net>

* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstxmptag.c:
* gst-libs/gst/tag/tag.h:
* tests/check/libs/tag.c:
* win32/common/libgsttag.def:
  tags: add basic xmp metadata support
  XMP metadata can be embedded in many media container formats. Implement own
  parser and formatter that can be used to convert between an xpacket and a
  GstTagList. Add unit tests.

2010-02-19 14:38:36 +0200  Stefan Kost <ensonic@users.sf.net>

* tests/icles/.gitignore:
* tests/icles/Makefile.am:
* tests/icles/test-xoverlay.c:
  example: add an example for xoverlay::set_render_rectangle()
  This add a new example which animates a target recangle for the video.

2010-02-19 14:46:43 +0200  Stefan Kost <ensonic@users.sf.net>

* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
  xvimagesink: implement set_render_rectangle
  Previously we hardcoded the target rectangle passes to Xv(Shm)PutImage. Extend
  the implementation to use a full rectangle and don't assume 0,0 for top,left.

2010-02-17 15:00:13 +0200  Stefan Kost <ensonic@users.sf.net>

* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/interfaces/xoverlay.h:
  xoverlay: add new vmethod ::set_render_rectangle()
  Add set_render_rectangle() vmethod to the interface to better support windowless
  toolkits (e.g. qt graphicsview or video on canvas in general). Right now we
  always fill the widget to 100%. With the patch we can use a rectangular target
  region. Fixes #610249.
  API: GstXOverlay::set_render_rectangle()

2010-02-16 12:06:08 +0200  Stefan Kost <ensonic@users.sf.net>

* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
  x(v)imagesink: take new size from event thread and do not poll for every frame
  We can update the geometry in ConfigureNotify (unless we disable event-
  handling). If event handling is disabled, one should use _expose() to trigger a
  redraw and update the geometry.

2010-03-10 21:51:59 +0100  Benjamin Otte <otte@redhat.com>

* common:
  Automatic update of common submodule
  From 9720a7d to df8a7c8

2010-03-10 21:01:20 +0100  Benjamin Otte <otte@redhat.com>

* configure.ac:
  Update for recent changes to common submodule
  This just replaces every "$ERROR_CFLAGS" usage with a usage of
  "$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as
  previously.
  Actually using that separation will happen later.

2010-03-10 20:43:46 +0100  Benjamin Otte <otte@redhat.com>

* common:
  Automatic update of common submodule
  From 0b6e072 to 9720a7d

2010-03-10 16:09:45 +0100  Benjamin Otte <otte@redhat.com>

* common:
  Automatic update of common submodule
  From 7cc5eb4 to 0b6e072

2010-03-10 14:36:34 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst-libs/gst/tag/gsttagdemux.c:
  tagdemux: do not cache FLUSH_START/_STOP events
  ... and similarly so for serialized events.

2010-03-10 14:34:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: provide correct error message if configured audio/video sink fails

2010-03-10 10:22:47 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* ext/vorbis/gstvorbisdec.h:
  vorbisdec: remove unused field

2010-02-02 11:34:10 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* tests/check/pipelines/vorbisdec.c:
  tests: enable strict discontinuity checking on vorbisdec pipeline
  Closes #423086.

2010-03-10 01:09:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 7aa65b5 to 7cc5eb4

2010-03-10 01:07:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/video/video.c:
  docs: fix Returns: for gst_video_parse_caps_color_matrix()

2010-03-10 00:46:34 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-03-10 00:42:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/check/elements/videorate.c:
  tests: fix typo in videorate unit test pipeline description
  Two consecutive ! ! leave a 'Link without source' error in the debug log.

2010-03-10 00:41:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/check/elements/videorate.c:
  tests: don't use deprecated functions in videorate unit test

2010-03-10 00:29:21 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* win32/common/libgstvideo.def:
  win32: add new API to libgstvideo.def

2010-03-09 15:39:55 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggmux.c:
  oggmux: Don't flush after every frame for theora

2010-03-09 21:26:58 +0000  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 44ecce7 to 7aa65b5

2010-03-09 13:05:23 -0800  David Schleef <ds@schleef.org>

* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
  video: Add color-matrix handling to caps

2010-01-30 22:55:01 -0800  David Schleef <ds@schleef.org>

* gst/videotestsrc/gstvideotestsrc.c:
  videotestsrc: Add color-matrix to caps

2010-02-26 16:25:59 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* pkgconfig/Makefile.am:
* tests/examples/overlay/Makefile.am:
* tools/Makefile.am:
  build: Make some more rules silent if requested

2010-02-26 15:40:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* configure.ac:
  configure: Use automake 1.11 silent rules instead of shave if available
  This makes sure that we use something that is still maintained and
  also brings back libtool 1.5 support.

2010-02-23 19:12:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Don't fail if there are subtitles and audio but no video
  Change playbin2 to not error out if there are subtitles and audio
  but no video. If visualizations are enabled the subtitles are rendered on top
  of the visualization stream, otherwise the subtitles are not linked at all and
  only the audio is played (and a warning message is posted).
  If there are only subtitles but neither audio nor video an error message is
  still posted.
  Fixes bug #610866.

2010-02-17 19:18:29 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
  volume: If a controller is used, use sample accurate property values
  Fixes bug #609801.

2010-03-09 19:17:04 +0100  Benjamin Otte <otte@redhat.com>

* gst-libs/gst/video/video.c:
  gstvideo: Fix typos in comments

2010-03-09 17:32:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* win32/common/_stdint.h:
* win32/common/config.h:
  Back to development

=== release 0.10.28 ===

2010-03-08 23:20:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
  Release 0.10.28

2010-03-08 23:19:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-03-08 21:57:03 +0100  Benjamin Otte <otte@redhat.com>

* ext/theora/gsttheoraenc.c:
  theora: Fix SIGFPE when using 0/1 framerate
  libtheora crashes with a 0 framerate, so let's forbid it.
  https://bugzilla.redhat.com/show_bug.cgi?id=571289

2010-03-08 14:50:25 +0000  David Schleef <ds@schleef.org>

* ext/ogg/dirac_parse.c:
  oggdemux: fix dirac header parsing
  Fixes #611900.

2010-03-08 14:46:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/overlay/Makefile.am:
  examples: make sure to dist qtgv-xoverlay.h header file
  This time for real.
  Fixes #610832.

2010-03-08 12:11:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpdepayload.c:
  basedepay: clarify some documentation

2010-03-08 11:25:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/alsa/gstalsasrc.c:
  alsasrc: return right number of bytes that we wrote

2010-03-08 11:20:51 +0100  Dake Gu <gudake@gmail.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtspconnection: fix handling of x-server-ip-address
  Fix handling of x-server-ip-address.

2010-03-02 11:25:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* docs/design/draft-keyframe-force.txt:
  docs: update keyframe force event
  Add field to send all headers.

=== release 0.10.27 ===

2010-03-06 00:09:29 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
  Release 0.10.27

2010-03-06 00:08:23 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-03-05 15:58:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  configure: first check for QtGui >= 4.6, only then for >= 4.0
  If we first check for >= 4.0 the second check for >= 4.6 will just
  short-cut since we are using the same prefix for the variables for
  both checks, and they've already been set previously. So the examples
  requiring >= 4.6 were built even in the >= 4.0 case.

2010-03-03 20:18:16 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
  0.10.26.4 pre-release

2010-03-03 20:17:31 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* po/ja.po:
  po: update translations

2010-03-03 20:15:44 +0000  Josep Torra Valles <n770galaxy@gmail.com>

* gst/playback/gstplaysink.c:
  playsink: avoid g_object_set() on NULL pointers
  There may not be an overlay element if a text-sink is set.
  Fixes #611702.

2010-03-01 12:17:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  oggstream: mark skeleton streams correctly
  Mark skeleton streams because we need to ignore them for calculating the
  duration of the stream.
  Fixes #611227

2010-02-24 01:10:09 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* po/nl.po:
* win32/common/_stdint.h:
* win32/common/config.h:
  0.10.26.3 pre-release

2010-02-23 16:57:53 +0100  Götz Waschk <waschk@mandriva.org>

* tests/examples/overlay/Makefile.am:
  examples: Dist header file for the Qt graphics view example
  Fixes bug #610832.

2010-02-23 11:41:20 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: use the chain begin_time instead of our counter
  We update the passed begintime argument to narrow our search region in the
  binary search. This means that it does not always contain the chain begin time
  after a couple of bisects. Use the real chain->begin_time to bring the
  granuletime to the time in the chain instead.
  Fixes #610005

2010-02-19 18:24:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* tests/check/elements/videorate.c:
  videorate: tests: New unit tests for upstream caps nego
  Adds unit tests that check videorate's upstream caps
  negotiation works properly (put passthrough caps
  first)
  Fixes #608025

2010-01-27 15:07:47 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/videorate/gstvideorate.c:
  videorate: Improve upstream negotiation
  Put peer pad caps preferred framerates first, indicating
  they are videorate's first choices, removing an unnecessary
  conversion.
  Fixes #608025

2010-02-21 19:52:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
  playbin2, playsink, subtitleoverlay: Set subtitle encoding properly
  For this add subtitle encoding properties to playsink and subtitleoverlay
  and update the values in the containing elements.
  Also update the font description in textoverlay or the used renderer
  element if it is changed during playback.
  Fixes bug #610310.

2010-02-22 13:01:19 +0200  Stefan Kost <ensonic@users.sf.net>

* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.cpp:
  examples: also add sink detection and set title to qt examples
  Also set a title in the qt examples like it is now done in the gtk example.
  Fix the newly added find_video_sink in the gtk example and add similar function
  to the qt examples.

2010-02-19 14:40:43 +0200  Stefan Kost <ensonic@users.sf.net>

* tests/examples/overlay/.gitignore:
  gitignore: ignore files in new example directroy

2010-02-17 14:59:33 +0200  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/video/Makefile.am:
  make: fix copy and paste error in git rules (audio<->video)

2010-02-19 17:44:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Ghost the video sinkpad if a text sinkpad is available
  Only don't ghost it if no visualizations are need and if
  no text is needed and no textchain was created yet.
  Fixes bug #610379.

2010-02-19 00:22:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
  0.10.26.2 pre-release

2010-02-19 00:20:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-02-19 00:17:51 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/overlay/.gitignore:
  Ignore new overlay examples

2010-02-18 23:47:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/overlay/gtk-xoverlay.c:
  examples: don't hard-code xvimagesink for Gtk+ GstXOverlay example
  Try to find a working videosink, don't hardcode xvimagesink. Also
  add some borders to window and give it a title so that it's clear
  that this is really a Gtk+ window and not a window created by the
  videosink.

2010-02-18 11:42:55 -0800  David Schleef <ds@schleef.org>

* gst/tcp/gsttcp.c:
  tcp(client/server)src: Fix handling of closed sockets
  The peer closing the socket should cause an EOS, instead of
  silently doing nothing.  This changes the behavior to be
  more like fdsrc.  Fixes: #610386

2010-02-18 12:42:53 +0000  Patrick Radizi <patrick.radizi@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtspconnection: make sure not to dereference NULL username or password
  Fixes #610268.

2010-02-17 21:22:54 -0800  David Schleef <ds@schleef.org>

* ext/theora/gsttheoradec.c:
  theoradec: Fix chroma copying for 4:2:2
  Fix mixup of height/width, causing only half the chroma lines to
  be copied when outputting buffers.  Fixes: #610329.

2010-02-16 15:43:26 +0200  Stefan Kost <ensonic@users.sf.net>

* configure.ac:
* gst-libs/gst/interfaces/xoverlay.c:
* tests/examples/Makefile.am:
* tests/examples/overlay/Makefile.am:
* tests/examples/overlay/gtk-xoverlay.c:
* tests/examples/overlay/qt-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.cpp:
* tests/examples/overlay/qtgv-xoverlay.h:
  examples: add video overlay examples for gtk, qt and qt graphics view
  Add simple videotestsrc ! xvimagesink examples using gtk and qt. This patch also
  adds all boilerplate to configure for using c++. The qt based examples are
  optional like their gtk counterparts.

2010-02-16 17:20:01 +0200  Stefan Kost <ensonic@users.sf.net>

* docs/libs/compiling.sgml:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
  docs: cleanup library docs
  Correct name of included files. Remove files that are not used anymore. Add many
  new api entries to their sections.

2010-02-15 11:11:04 +0200  Stefan Kost <ensonic@users.sf.net>

* tests/icles/test-colorkey.c:
  test-colorkey: remove the XInitThreads()
  We don't do this is any other example, this should be done for us in gdk it if
  would be needed.

2010-02-16 10:09:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: use same message string for missing elements as in playbin
  Use the same translated message string for missing core elements as
  playbin uses, which is a bit nicer and also indicates that there is
  something wrong with the user's GStreamer installation (which arguably
  is the case if elements like typefind or queue2 are missing).

2010-02-08 13:54:57 +0200  Kaj-Michael Lang <milang@tal.org>

* gst/typefind/gsttypefindfunctions.c:
  typefind: Handle stm module format
  Fixes #609314.

2010-02-15 12:10:10 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* ext/vorbis/gstivorbisdec.c:
  ivorbisdec: set rank to SECONDARY

2010-02-15 12:09:53 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* configure.ac:
* ext/Makefile.am:
* ext/vorbis/Makefile.am:
* ext/vorbis/gstivorbisdec.c:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
* ext/vorbis/gstvorbisdeclib.c:
* ext/vorbis/gstvorbisdeclib.h:
  vorbisdec: also support ivorbis tremor decoder
  ... which only needs a bit of refactoring and extracting to support
  the minor difference in (i)vorbis interface.
  Fixes #609063.

2010-02-03 14:37:43 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
  vorbisdec: reduce some hard-coding
  ... such as assuming float all over, and base src caps on template caps.

2010-02-15 10:23:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/elements/playbin.c:
  playbin: Fix the primary-decoder-missing test with USE_DECODEBIN2

2010-02-15 09:04:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggparse.c:
  oggparse: Fix another format string compiler warning

2010-02-15 08:56:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Fix format string compiler warnings

2010-02-15 08:48:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Post a missing element message and an error message if no uridecodebin can be found

2010-02-15 08:46:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Post missing element messages if a core plugin is missing
  And post a warning in cases where we can still continue to work
  or an error when the missing element is fatal.

2010-02-15 08:28:24 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/elements/playbin2.c:
  playbin2: Enable all unit tests
  They're all working and valgrind clean now.

2010-02-15 08:26:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: First post a missing-plugin message, then emit the unkown-type signal
  This makes sure that there *always* is a missing plugin message in the bus
  before any errors or warning messages.

2010-02-15 08:20:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Missing decoder errors should be STREAM CODEC_NOT_FOUND
  and not CORE MISSING_PLUGIN.

2010-02-15 08:18:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Free the subtitle URI

2010-02-15 08:06:44 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Post missing plugin messages if a required element can't be created
  Especially if no suitable URI source can be found.

2010-02-15 06:50:29 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/elements/.gitignore:
  tests: Add decodebin2 test to .gitignore

2010-02-15 01:18:55 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Set ghostpad targets to NULL when freeing a decode chain
  Otherwise the ghostpad will still be linked to the peer and there
  will still be a reference kept, leading to nothing being unlinked
  and destroyed until decodebin2 is finalized.
  This fixes reuse of decodebin2 if a raw stream is connected to
  its sinkpad.

2010-02-15 01:17:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/Makefile.am:
* tests/check/elements/decodebin2.c:
  decodebin2: Add simple unit test, mainly a copy of the decodebin unit test
  The only difference between the two unit tests right now is,
  that the decodebin2 test resets the element to READY before trying
  to reuse it instead of NULL. decodebin2 guarantees to be reusable
  without going back to NULL.

2010-02-15 00:11:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  ogg: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
  See #609252.

2010-02-14 23:16:32 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 96dc793 to 44ecce7

2010-02-14 23:10:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/Makefile.am:
  playbin2: Enable playbin2 unit test
  It now contains a single working unit test and can be enabled.
  The other more useful unit tests still need fixing.

2010-02-14 22:16:31 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/elements/playbin.c:
  playbin: Fix indention in the unit test

2010-02-13 01:08:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/volume/gstvolume.c:
  volume: Replace this variables by self

2010-02-12 19:43:13 +0100  Josep Torra Valles <n770galaxy@gmail.com>

* gst/playback/gstplaysink.c:
  playsink: Reset the sink's state to NULL before unreffing it unless it's the same instance again
  This makes sure that we don't destroy the last reference before the
  element gets back to NULL state. Fixes assertion failures if a playbin2
  instance is reused but different sinks are automatically chosen because
  of different caps.

2010-02-12 18:00:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: fix Since tag

2010-02-12 14:19:33 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/riff/riff-read.c:
  riff: treat JUNQ chunks like JUNK chunks

2010-02-12 14:29:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: Update basesrc segment duration and post duration messages from the streaming thread

2010-02-11 14:10:02 +0200  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/tag/tags.c:
  tags: improve docs about determining the encoding

2010-02-11 14:09:05 +0200  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/tag/gstvorbistag.c:
  comment: fix wrong header comment

2010-02-01 13:50:14 +0200  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/riff/riff-ids.h:
  riff: add a variant of the JUNK tag that several adobe products produce
  JUNQ has same semantics as JUNK.

2010-02-01 19:01:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: add min-percent property
  Emit need-data when the amount of data in the internal queue drops below
  min-percent.
  Fixes #608309

2010-02-01 18:56:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: cleanups
  Avoid some typechecks.
  Avoid dereferencing appsrc->priv all the time.

2010-02-01 18:55:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/app/gstappsink.c:
  appsink: cleanups
  Avoid some typecasting.
  Avoid dereferencing appsink->priv all the time.

2010-02-01 15:09:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: avoid some typecasts

2010-01-29 16:34:44 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: ignore \n and \r as the first line
  Be more forgiving for bad servers and ignore \r and \n when we are looking for
  the response/request line.
  See #608417

2010-02-10 16:05:29 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: fail gracefully on bad Content-Length headers
  Be careful when allocating the amount of bytes specified in the Content-Length
  because it can be an insanely huge value. Try to allocate the memory but fail
  gracefully with a nice error when the allocation failed.

2010-02-10 10:12:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
  ffmpegcolorspace: Add conversions from all ARGB formats to AYUV and back

2010-02-09 17:39:21 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: Update segment duration and post a duration message if the duration changes
  Fixes bug #609423.

2010-02-11 10:56:17 +0100  Benjamin Otte <otte@redhat.com>

* tests/examples/seek/Makefile.am:
  build: link to libm in examples that use it
  This fixes build failure in Fedora 13.

2010-02-11 01:11:30 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* MAINTAINERS:
  Update MAINTAINERS, add myself

2010-02-11 23:57:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  configure: back to development
  Slushy freeze remains in effect.

=== release 0.10.26 ===

2010-02-10 20:17:36 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
  Release 0.10.26

2010-02-10 20:16:37 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-02-08 11:21:35 +0100  Benjamin M. Schwartz <bens@alum.mit.edu>

* ext/theora/gsttheoradec.c:
  theoradec: PARs of 0:x, x:0 and 0:0 are all allowed and map to 1:1
  Fixes #609252.

2010-01-24 12:31:04 +0000  Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>

* ext/ogg/gstoggstream.c:
  oggdemux: use the default granpos functions for kate streams
  Set timestamps on kate packets. See bug #600929.

2010-02-05 01:18:43 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* win32/common/_stdint.h:
* win32/common/config.h:
  0.10.25.3 pre-release

2010-02-04 18:52:59 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* po/bg.po:
  po: update translations

2010-02-04 18:32:48 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  Revert "playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler"
  This reverts commit 7335ce5d3e03c126a417a721571cb6f3af136ecf.
  Support abusing the uri property to configure the next uri to play
  outside of the about-to-finish handler for the time being after all.
  We also shouldn't use thread private structures for this, since it
  should be possible to block the thread that emitted about-to-finish
  while the main thread sets the uri property. See #607226.

2010-02-02 10:18:05 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Don't leak allocated buffers
  This can happen if the combined flow return is not OK although the
  allocation succeeded or if the packet in question is a BOS and we're
  not going to push headers.
  Fixes bug #608699.

2010-02-01 11:44:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: clean up decodebin properties
  When reusing a decodebin2 element, clear the properties we might have changed,
  to their default values or else we might end up with old configuration.
  Fixes #608484

2010-01-29 13:56:05 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: when no uri is set, post an error message
  When no uri is set, don't just return STATE_CHANGE_FAILURE from the
  state change function, but actually post an error message.

2010-01-30 15:18:13 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 15d47a6 to 96dc793

2010-01-28 17:12:34 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/adder/gstadder.c:
  adder: don't hold object lock when calling peer elements
  Do not hold the object lock while we call methods on peer elements as this can
  lead to deadlocks.
  Fixes #608179

2010-01-27 01:12:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  0.10.25.2 pre-release

2010-01-27 01:07:55 +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/interfaces-enumtypes.h:
* win32/common/pbutils-enumtypes.c:
* win32/common/video-enumtypes.c:
  win32: update generated files for non-autotools win32 builds

2010-01-27 00:56:00 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

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

2010-01-27 00:41:24 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/audio/gstaudiosrc.c:
  audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type

2010-01-26 16:47:40 +0100  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstdecodebin2.c:
  decodebin2: Don't skip an element when getting the topology
  Fixes #608167

2010-01-24 14:41:44 +0000  Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>

* ext/ogg/gstoggdemux.c:
  oggdemux: sparse streams aren't timed by end time, and their duration isn't implicit
  Fixes timestamps and durations on Kate subtitle streams.
  See http://www.xiph.org/ogg/doc/ogg-multiplex.html section 'start-time and
  end-time positioning' for some more details, and bug #600929.

2010-01-23 20:15:08 +0000  Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>

* ext/ogg/gstoggstream.c:
  oggdemux: properly set up the media type for kate streams
  See #600929.

2010-01-25 18:57:52 +0100  Julien Moutte <julien@fluendo.com>

* gst/playback/gstsubtitleoverlay.c:
  subtitleoverlay: relax caps template on sink pads
  Allow any caps on sink pad templates as we could do passthrough with non raw
  video caps.

2010-01-25 15:14:56 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.h:
  oggdemux: use right type for the serialno
  Use a consistent type for the serialno to avoid problems when comparing between
  signed and unsigned variants.
  Fixes #607926

2010-01-25 14:00:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: don't push headers twice
  Don't push the stream headers twice but only in the activation of a chain.
  Fixes #607929

2010-01-25 13:18:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

  Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base

2010-01-25 12:31:24 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
  oggdemux: rename a variable
  Rename the 'seekable' variable to 'pullmode'. We might be able to seek in push
  mode too eventually.

2010-01-25 12:22:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstinputselector.c:
  Revert "inputselector: Protect g_object_notify() with the object's mutex"
  This reverts commit a37426c41c80fd21e5017fea01a786c05bcd9661, it's
  causing deadlocks with playbin2.

2010-01-24 20:55:26 +0100  Kipp Cannon <kcannon@ligo.caltech.edu>

* gst/playback/gstinputselector.c:
  inputselector: Protect g_object_notify() with the object's mutex
  This works around the thread unsafety of g_object_notify()
  Fixes bug #607513.

2010-01-24 20:46:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Add typefinder for ISO MP4 files
  Fixes bug #607848.

2010-01-24 13:29:07 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: fix crash when freeing headers
  Use _ogg_packet_free() instead of gst_mini_object_unref in one more
  place now that the header list contains ogg packets and not buffers.
  file: Stephen_Fry-Happy_Birthday_GNU-nq_600px_425kbit.ogv

2010-01-24 08:57:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Strip trailing \0 for subtitle OGM streams
  Fixes bug #607870.

2010-01-23 22:09:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Correctly set DELTA_UNIT flag for OGM streams

2010-01-23 22:05:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Don't strip all 0-bytes from the end of OGM packets
  This fixes broken packets pushed downstream by oggdemux for
  MPEG4 streams for example.

2010-01-23 22:03:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Extract tags from OGM text streams and don't push them downstream

2010-01-23 14:46:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Store header/queued packets as ogg_packet and use normal peer chaining functions to pass them downstream

2010-01-23 15:25:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefinding: optimise AC-3 typefinder a bit
  Make AC-3 typefinder use the DataScanCtx stuff so we don't have to
  do gst_type_find_peek() in the inner loop all the time. Also return
  when we've suggested AC3 caps, instead of continuing with the loop.

2010-01-23 14:31:15 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  Revert "typefind: Reduce number of calls to gst_type_find_peek."
  This reverts commit c661bfaa991c58f1fbd9fbc0dae90b8b2c27f92b.
  This breaks AC-3 typefinding for all cases where the first frame
  is at an offset > 0.

2010-01-23 15:35:05 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/pbutils/descriptions.c:
  pbutils: Add description for Zip Block Motion Video

2010-01-23 15:34:54 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: Add mapping for Zip Block Motion Video

2010-01-23 15:26:37 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: YUNV is a fourcc which is also used for YUY2 raw video

2010-01-23 15:13:45 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: vp61 and VP61 are also valid On2 VP6 fourcc

2010-01-23 15:10:45 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: Add mapping for On2 VP5

2010-01-23 15:04:35 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: Add mapping for Sigma-Designs MPEG4
  It's actually a xvid-compatible stream. both xviddec and ffmpeg handle it.

2010-01-23 14:35:28 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/pbutils/descriptions.c:
  pbutils: Add description for LOCO Lossless codec

2010-01-23 14:35:16 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: Add mapping for LOCO Lossless codec

2010-01-23 14:08:39 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: Add support for YV12 / Uncompressed packed YVU 4:2:2

2010-01-23 13:50:26 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/pbutils/descriptions.c:
  pbutils: add description for Autodesk Animator codec

2010-01-23 13:50:09 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/riff/riff-media.c:
  riff: Add mapping for Autodesk Animator Codec

2010-01-23 13:20:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: ...and set caps on queued packet buffers too

2010-01-23 13:19:08 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Set caps on header buffers

2010-01-22 16:23:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: handle raw sources about-to-finish signals
  When we are dealing with a source that produces raw audio/video, we don't use a
  decodebin2 to decode the data and we thus don't have the drained/about-to-finish
  signal emited. To fix this, we add a padprobe on the source pads and emit the
  drained signal ourselves. This then makes playbin2 emit the about-to-finish
  signal for raw sources such as cdda://
  Fixes #607116

2010-01-22 16:15:54 +0200  Stefan Kost <ensonic@users.sf.net>

* gst/typefind/gsttypefindfunctions.c:
  typefind: include stdio.h for sscanf

2010-01-22 01:49:38 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefinding: add PNM typefinder
  Add PNM typefinder, so we can remove the one that's in the PNM plugin
  in -bad (which btw uses different/wrong media types that don't match
  the ones used by gdkpixbufdec) and people don't make fun of us for
  loading image decoders when typefinding and playing back audio files.

2010-01-21 19:31:23 +0100  Thijs Vermeir <thijsvermeir@gmail.com>

* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
  ffmpegcolorspace: rename performance category
  rename the performance category to ffmpegcolorspace_performance
  as there is already a global GST_CAT_PERFORMANCE in core

2010-01-21 17:32:33 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
  oggdemux: keep track of added pads
  Keep track of the pads we added and removed.
  Remove some unused fields.
  Don't add pads for which we don't have caps.

2010-01-21 17:31:13 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  oggstream: don't call NULL setup functions
  If we find a known mapper but it doesn't have a setup function, simply skip it
  instead of crashing.

2010-01-21 17:30:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  oggstream: avoid division by 0 on bad annodex streams

2010-01-21 13:47:01 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/pbutils/descriptions.c:
  pbutils: Add description for y4m container

2010-01-19 14:31:34 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  basertppayload: ptime/maxptime should be unsigned
  https://bugzilla.gnome.org/show_bug.cgi?id=607403

2010-01-18 21:16:32 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
  basertppayload: ptime should be in nanoseconds
  https://bugzilla.gnome.org/show_bug.cgi?id=607403

2010-01-20 00:53:20 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 14cec89 to 15d47a6

2010-01-19 13:33:06 -0800  David Schleef <ds@schleef.org>

* gst/typefind/gsttypefindfunctions.c:
  typefind: rewrite h.264 detection
  Make detection simpler: check for NALs, check that they make
  sense, and report how certain we are that it's a raw H.264 stream.
  Fixes: #583376.

2010-01-18 14:33:30 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  basertppayload: Reject empty caps
  https://bugzilla.gnome.org/show_bug.cgi?id=607353

2010-01-19 08:39:14 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: No need to subtract begin time
  Last stop is already based on the chain start and there is no need
  to subtract the chain start as it may lead to a negative overflow.
  This was causing seeking issues when the target chain was not
  the first one (that has chain start = 0)
  Fixes #606382

2010-01-19 09:25:35 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/audio/audio.h:
  audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
  Fixes bug #607381.

2010-01-18 15:22:52 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: granulepos is relative to its chain
  When performing seeks, the granulepos should be offset by
  its chain start time to avoid using wrong values to
  update segment's last_stop. A sample file is indicated on
  bug #606382

2010-01-18 17:57:16 +0100  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/pbutils/descriptions.c:
  pbutils: Add description for MXF container format

2010-01-18 10:07:30 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: re-use iterator callback to avoid code duplication

2010-01-18 02:08:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: when looking for sink properties, make sure they have the right type
  We don't want to end up setting values on elements where the property is of
  a different type than we expect. Can't transform the value either, since we
  can't really make assumptions about the scale and transform function.
  Fixes crashes when using playbin2 with apexsink (#606949).

2010-01-18 09:30:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler
  Changing the URIs in a state > READY results in unexpected behaviour,
  i.e. the new URIs are only used after the current track has finished.
  Fixes bug #607226.

2010-01-15 19:52:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: sprinkle some more locking
  ... to avoid races and ensure some data structure consistency.
  See also #574289.

2010-01-14 18:26:03 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: mind blocked pads when shutting down
  Fix regression in shutdown deadlock handling now that the
  target of a ghostpad is blocked instead of ghostpad itself.
  See also #574293.

2010-01-14 13:36:23 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Fix disabling of subtitles if subtitles were used before
  In this case the video still goes through the text chain and
  subtitles are still going in there, in case subtitles are
  enabled again. This makes sure that re-enabling subtitles
  happens instantly.
  Fixes hanging video when disabling subtitles, caused by an
  unliked video pad.

2010-01-14 10:43:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: fix pad ref leak

2010-01-12 21:42:59 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* docs/plugins/Makefile.am:
  docs: fix out-of-source build

2009-04-29 11:50:03 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* tests/icles/stress-playbin.c:
  stress-playbin: fix error return check

2010-01-14 10:10:23 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/theora/Makefile.am:
* ext/theora/gsttheora.c:
* ext/theora/gsttheoradec.c:
* ext/theora/gsttheoraenc.c:
* ext/theora/gsttheoraparse.c:
* ext/theora/theora.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c:
  theora: Rename source files to have the same name as the headers

2010-01-14 10:07:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbis.c:
* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisparse.c:
* ext/vorbis/gstvorbistag.c:
* ext/vorbis/vorbis.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
  vorbis: Rename source files to have the same name as the headers

2010-01-14 10:05:35 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/vorbis/Makefile.am:
* ext/vorbis/gstvorbiscommon.c:
* ext/vorbis/gstvorbiscommon.h:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
  vorbis: Move channel layout definitions into a single separate file
  ...instead of having two copies.

2010-01-14 08:19:55 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
  vorbis: Add official 6.1 and 7.1 channel mappings
  These are in the Vorbis spec since 2010-01-13. Fixes bug #606926.

2010-01-13 23:05:45 +0100  Benjamin Otte <otte@redhat.com>

* gst-libs/gst/rtsp/gstrtspdefs.c:
  rtsp: Don't define h_error ourselves
  It's included from netdb.h and that header might define it differently,
  which can lead to build failures.

2010-01-13 17:36:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefind: mp4 video is not parsed

2010-01-13 12:49:20 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefind: Add aac stream-format to caps
  Also add the aac stream-format field on the caps when
  detecting it.

2010-01-13 09:39:54 +0100  Brijesh Singh <brijesh.ksingh@gmail.com>

* gst/playback/gstplaysink.c:
  playsink: Fix handling of the native audio/video flags
  Fixes bug #606687.

2010-01-12 16:35:50 +0100  Edward Hervey <bilboed@bilboed.com>

* ext/ogg/gstoggdemux.c:
  oggdemux: Fix unitialized variable.
  If the package isn't handled, gracefully return GST_FLOW_OK.

2010-01-10 23:50:02 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/interfaces/xoverlay.c:
  docs: flesh out GtkXOverlay docs some more and add example for Gtk+ >= 2.18
  Explain why the whole bus sync handler mess is needed. Add section about
  how to use GstXOverlay in connection with Gtk+ and mention the Gtk+ API
  break issue and how to work around it (see #601809).

2010-01-10 21:18:04 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/netbuffer/gstnetbuffer.c:
  docs: minor netbuffer documentation fix

2010-01-10 20:41:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
  po: update translated strings
  Queue2 moved into core, so remove its strings.

2010-01-08 16:57:40 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.h:
  oggdemux: push headers when activating chains
  Keep a list of headers for each stream of a chain. When a chain is activated,
  push the headers before pushing the data so that decoders can sync.
  Fix seeking in chains, take the chain start time into account when comparing
  timestamps.
  See #606382

2010-01-07 15:26:57 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/lang-tables.dat:
* gst-libs/gst/tag/lang.c:
  tag: fix up disting of lang-tables.c more correctly
  lang-tables.c is included by lang.c and not really a proper source
  file that should be compiled into its own object, so rename it to
  lang-tables.dat and put it into EXTRA_DIST instead to ensure it
  gets disted.

2010-01-07 13:50:03 +0000  Christian Schaller <christian.schaller@collabora.co.uk>

* gst-libs/gst/tag/Makefile.am:
* gst-plugins-base.spec.in:
  Add missing source file for tagger to Makefile and update spec file

2010-01-06 18:30:57 -0800  Mark Yen <mook@songbirdnest.com>

* gst-libs/gst/riff/riff-media.c:
  riff-media: handle 32 bit raw RGB video.

2010-01-06 13:57:51 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  oggdemux: decide flac header packet by content rather than count

2010-01-06 13:56:26 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: reset header packet count at bos page

2010-01-06 13:39:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiopayload: add support for buffer-lists

2010-01-06 11:33:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

  Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base

2010-01-05 17:17:58 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Ignore zero framerate
  https://bugzilla.gnome.org/show_bug.cgi?id=606163

2009-12-29 18:45:32 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  basertpaudiopayload: Respect ptime if it is given
  If the ptime is given in the caps, respect it and force the minimum
  and maximum sizes to be exactly the requested ptime.
  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-29 18:36:29 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
  rtpbasepayload: Store ptime from caps
  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2009-12-02 19:40:58 +0530  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  basertppayload: Accept maxptime from caps
  https://bugzilla.gnome.org/show_bug.cgi?id=606050

2010-01-05 14:11:06 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  oggdemux: enhance flac packet duration calculation

2010-01-05 10:38:41 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

  Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base

2010-01-04 09:49:25 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/seek/seek.c:
* tests/icles/test-colorkey.c:
  examples: use Gtk+-2.18 API conditionally
  so the seek example and colorkey test work with older Gtk+ versions
  as well.
  Fixes #605960.

2009-12-29 00:53:53 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/icles/test-colorkey.c:
  tests: fix colorkey test up for Gtk+ >= 2.18
  Make test-colorkey work with newer versions of Gtk+.
  See #601809.

2009-12-29 00:40:27 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/seek/seek.c:
  examples: make seek example work with Gtk+ >= 2.18
  Gtk+ broke API slightly with the introduction of
  client-side windows in Gtk+ 2.18. Fix up seek
  example to work with newer Gtk+ versions.
  Fixes #601809.

2009-12-26 23:29:24 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/icles/stress-xoverlay.c:
  tests: fix warning and memory leak in stress-overlay test
  Not all messages have structures and we need to unref messages
  when returning GST_BUS_DROP in the sync bus handler.

2009-12-26 18:46:50 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
  audiorate: correctly eat empty and dummy buffers

2009-12-24 19:56:55 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/adder/gstadder.c:
  adder: be a lot smarter with buffer management
  Detect EOS faster.
  Try to reuse one of the input buffer as the output buffer. This usually works
  and avoids an allocation and a memcpy.
  Be smarter with GAP buffers so that they don't get mixed or cleared at all. Also
  try to use a GAP buffer as the output buffer when all input buffers are GAP
  buffers.

2009-12-24 16:30:23 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/adder/Makefile.am:
* gst/adder/gstadder.c:
* tests/check/elements/adder.c:
  adder: use collectpads clipping function
  Install a clipping function in the collectpads and use the audio clipping helper
  function to perform clipping to the segment boundaries.
  Fixes #590265

2009-12-24 13:58:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/adder/gstadder.c:
  adder: fix juvenile comment

2009-12-23 21:24:48 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: fix typo in debug message

2009-12-23 18:18:03 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: avoid some type checks

2009-12-23 17:08:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: avoid leaking selector request pads

2009-12-23 15:46:25 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: avoid leaking queue and typefind
  Don't leak the queue and typefind elements that we might link after the
  source element.

2009-12-23 15:43:52 +0100  Jonathan Matthew <jonathan@d14n.org>

* gst/playback/gsturidecodebin.c:
  uridecodebin: don't name the queue
  There is no reason to name the queue.
  Fixes #605219

2009-12-23 15:30:50 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* win32/common/libgstrtp.def:
  defs: update defs with new symbols

2009-12-22 20:15:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
* gst-libs/gst/rtp/gstrtcpbuffer.h:
  rtcpbuffer: add helper functions for SDES types
  Add functions to convert SDES names to their types and back. Will be used later
  to set SDES items using a GstStructure.
  See #595265

2009-12-21 19:12:02 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 47cb23a to 14cec89

2009-12-21 18:45:58 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
  audiorate: add Since marker for the new tolerance property

2009-12-21 07:57:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/tag/lang.c:
  docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy

2009-12-21 07:50:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/examples/app/appsrc-ra.c:
* tests/examples/app/appsrc-seekable.c:
* tests/examples/app/appsrc-stream.c:
* tests/examples/app/appsrc-stream2.c:
  tests: don't use deprecated GLib API g_mapped_file_free
  Fixes #605100.

2009-12-20 17:34:46 -0800  David Schleef <ds@schleef.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theoraenc: Add encoder controls for libtheora 1.1
  Added drop-frames, cap-overflow, cap-underflow, and rate-buffer.

2009-12-19 21:40:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  baseaudiosink: increase default drift tolerance to fix glitches with WMA
  Increase default drift tolerance to 40ms to avoid glitches with decoders
  or formats where there's a lot of timestamp jitter for some reason or
  another (in this case: asf/wma), at least until we implement timestamp
  smoothing.

2009-12-16 11:43:27 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: add some debugging

2009-12-15 18:41:38 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
* gst/audiorate/gstaudiorate.h:
  audiorate: add a tolerance property
  It may not be uncommon for the input timestamps to experience some jitter
  around the 'perfect time'.  As such, instead of regularly adding and dropping
  samples, optionally allow for some tolerance in a more relaxed approach.
  API: GstAudioRate:tolerance

2009-12-15 19:50:56 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/audiorate/gstaudiorate.c:
  audiorate: add documentation

2009-12-15 16:52:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/Makefile.am:
* gst/audiorate/gstaudiorate.c:
* gst/audiorate/gstaudiorate.h:
  audiorate: use separate header file

2009-12-14 21:17:57 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
  audiorate: set DISCONT when resyncing (e.g. newsegment)

2009-12-14 18:47:27 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
  audiorate: also fill up segments if possible

2009-12-15 19:29:29 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
  audiorate: fix segment handling
  Do not compare a media (buffer) time to a (bogus) running time
  (or their offset equivalents).

2009-12-15 19:22:45 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/audiorate/gstaudiorate.c:
  audiorate: properly report truncated samples as dropped samples

2009-12-13 18:43:56 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/tag/lang.c:
  docs: mention that gst_tag_get_language_name() may return NULL

2009-12-13 18:42:11 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/check/libs/tag.c:
  checks: some more testing for the new language code functions

2009-12-12 18:58:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
  docs: misc. mixer docs improvements

2009-12-12 18:16:39 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
  docs: add short descriptions for API reference contents page

2009-12-12 17:43:26 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/mklangtables.c:
  tag: make internal language names table static

2009-12-12 17:41:44 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/tag/lang.c:
* gst-libs/gst/tag/mklangtables.c:
  tag: don't use GLib 2.22 API
  g_mapped_file_unref() was introduced in GLib 2.22, but we depend
  only on GLib 2.18, so use g_mapped_file_free() when compiling
  against older GLib versions until we bump the GLib dependency.

2009-12-11 23:59:54 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* .gitignore:
* configure.ac:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/lang-tables.c:
* gst-libs/gst/tag/lang.c:
* gst-libs/gst/tag/mklangtables.c:
* gst-libs/gst/tag/tag.h:
* tests/check/libs/tag.c:
* win32/common/libgsttag.def:
  tag: add some utility functions for language codes and tags
  Add some utility functions for language tags and ISO-639
  codes. These are useful for both GUIs and elements. The
  iso-codes package is used for language name translations
  if available.
  API: gst_tag_get_language_codes()
  API: gst_tag_get_language_name()
  API: gst_tag_get_language_code()
  API: gst_tag_get_language_code_iso_639_1()
  API: gst_tag_get_language_code_iso_639_2B()
  API: gst_tag_get_language_code_iso_639_2T()

2009-12-11 12:02:15 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  ogg: ogm video has constant packet duration

2009-12-10 22:47:53 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggstream.c:
  oggdemux: implement old fLaC mapping

2009-12-10 17:53:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/tcp/gsttcpclientsrc.c:
  tcpclientsrc: unset flushing state too
  When unlocking, we set the flushing state on the fdset. Implement unlock_stop so
  that we can use it to unset the flushing state again.
  Fixes #577326

2009-12-10 16:09:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
  oggdemux: remove redundant fields

2009-12-09 19:03:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/vorbis/gstvorbisdec.h:
* ext/vorbis/vorbisdec.c:
  vorbisdec: adapt to new oggdemux
  Remove all granulepos hacks and simply use the timestamps from the new oggdemux
  like any other decoder.

2009-12-09 19:04:18 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/vorbis/vorbisdec.c:
  vorbisdec: fix peer query

2009-12-09 17:24:01 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/theora/theoradec.c:
  theoradec: fix query

2009-12-09 16:55:52 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/theora/theoradec.c:
  theoradec: small cleanups

2009-12-09 16:38:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/vorbis/vorbisdec.c:
  vorbisdec: use gst_pad_peer_query()

2009-12-09 12:10:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: fix video when subtitles disabled
  When we have a source with subtitles but they were disabled with the flags,
  still ghostpad the video pad instead of leaving it unlinked.

2009-12-09 09:47:30 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Only flush downstream on seeks for flushing seeks

2009-12-09 09:35:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Proxy buffer allocation on the video sinkpad to the srcpad

2009-12-08 17:30:39 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* tests/examples/seek/seek.c:
  seek: update slider only 25 times a second
  don't update the slider a 100 times a second, it's likely higher than the screen
  framerate and just wastes cpu.

2009-12-08 17:23:53 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
  theora: remove granulepos hacks
  Remove the granulepos hacking now that oggdemux outputs timestamps like any
  other demuxer.

2009-12-08 13:40:18 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Fix stream-changed message list iteration
  When iterating the list and removing the current element, first
  get the next element and then remove the current one and not
  the other way around.

2009-12-07 18:49:43 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: improve keyframe seeking
  Improve keyframe seeking.
  Fix reverse playback.

2009-12-07 15:42:05 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: implement keyframe seeking
  Implement keyframe seeking in oggdemux by doing the double seek trick. First
  seek to the required position, then read pages for all streams to grab the
  granulepos (to know the timing of the keyframe) of each stream, then seek back
  to the first keyframe.

2009-12-07 09:13:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Some minor cleanup

2009-12-06 18:05:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Reset stream segments on FLUSH_STOP and don't adjust QoS events for non-time segments

2009-12-04 16:35:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: fix timestamps after seek
  After a seek, discard all packets before the packet with the granulepos on it so
  that the output buffers contain valid timestamps.
  Reorder some code so that we check the timestamps before allocating and pushing
  an output buffer.
  Do more checks on valid packets in ogm mode.

2009-12-04 15:39:59 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: add comment

2009-12-04 14:01:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: don't do math with invalid granulepos
  When the current granulepos is unknown and set to -1, don't try to add durations
  to it.

2009-12-04 13:14:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
  oggdemux: guard against wrong granulepos
  Clamp the initial granulepos to 0 instead of going negative for some badly muxed
  ogg files.

2009-12-04 12:26:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/theora/theoradec.c:
  theoradec: don't fail on bogus granulepos
  Do some additional checks on the granulpos timestamp before using it for
  calculating the duration because oggdemux generates wrong granulepos now.
  Fixes seeking somewhat again.

2009-12-03 20:05:29 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
  oggdemux: reimplement OGM support
  OGM demuxing no longer requires helper elements.  It's done internally
  in oggdemux.  Vorbis comments are still not handled because I don't
  have anything to test with.

2009-12-03 17:02:11 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggstream.c:
  oggdemux: fix for I-frame-only theora

2009-12-03 01:16:17 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  ogg: log when ogg mapper doesn't accept the setup header packet

2009-12-02 02:08:46 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  ogg: extract width, height and PAR from theora header and add to caps

2009-12-03 23:43:08 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggstream.c:
  ogg: extract number of channels from FLAC, speex and vorbis headers
  Because we can.

2009-12-03 22:14:34 +0200  Stefan Kost <ensonic@users.sf.net>

* gst/playback/gstplaybin2.c:
  build: fix build with debug logging disabled.

2009-12-03 21:07:49 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
  ogg: more print fixes
  gstoggstream.c:419: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘gint64’
  gstoggdemux.c:2253: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
  gstoggdemux.c:2333: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’

2009-12-03 16:57:48 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.c:
  ogg: Fixing some printf format strings
  Fixes some printf format strings to make it build on mac.

2009-12-03 18:08:49 +0200  Stefan Kost <ensonic@users.sf.net>

* gst/playback/gstfactorylists.c:
* gst/playback/gstfactorylists.h:
* gst/playback/gstplaybin2.c:
  playbin2: don't iterate the factory lists in non-debug mode
  When debugging is disabled, we won't see anything printed anyway.

2009-12-02 23:55:55 -0800  David Schleef <ds@schleef.org>

* gst/videoscale/vs_4tap.c:
  Build fix for MSVC

2009-12-02 23:27:55 +0200  Stefan Kost <ensonic@users.sf.net>

* gst/subparse/qttextparse.c:
  build: add missing includes for sprintf and atoi

2009-12-01 16:42:42 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/subparse/gstsubparse.c:
* gst/subparse/qttextparse.c:
  subparse: Add support for some tags of qttext
  Currently supporting timescale, timestamps, font, size,
  textColor, backColor, plain, bold and italic
  Fixes #603357

2009-12-01 13:13:24 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
* gst/subparse/qttextparse.c:
* gst/subparse/qttextparse.h:
  subparse: add qttext support
  Adds basic support for qttext subtitles, still lacks markup tags
  to make it prettier, but the plain text already works.
  Implemented according to:
  http://www.apple.com/quicktime/tutorials/texttracks.html
  http://www.apple.com/quicktime/tutorials/textdescriptors.html
  Fixes #603357

2009-12-01 13:22:57 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/subparse/gstsubparse.c:
  subparse: conditionally cleanup sami context
  Only cleanup sami context if we are parsing sami subtitles,
  otherwise we might have crashes.

2009-12-01 13:19:35 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/subparse/gstsubparse.c:
  subparse: Add missing caps to sink caps template
  Some caps were missing from the sink caps template when
  xml was disabled

2009-12-01 15:06:10 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 87bf428 to 47cb23a

2009-12-01 14:14:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* common:
  Automatic update of common submodule
  From da4c75c to 87bf428

2009-11-30 10:22:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
  subtitleoverlay: Fix some pad refcount issues
  Fixes bug #603345.

2009-11-27 18:54:57 +0100  Edward Hervey <bilboed@bilboed.com>

* common:
  Automatic update of common submodule
  From 53a2485 to da4c75c

2009-11-25 17:04:41 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
  oggdemux: handle theora streams with 0 keyoffset

2009-11-25 16:53:26 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
  oggdemux: Handle unknown streams

2009-11-26 14:30:33 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  Revert "textoverlay: First draw outline text and then the real text"
  This reverts commit 60aa09d28c1f9fd29b56876d7ac6c0366d6cef4d.
  First drawing the real text and then the outline produces ugly
  text in lower resolutions. The outline line width needs to be somehow
  changed relative to the resolution. Fixes bug #602924.

2009-11-26 10:30:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/audio/gstaudiofilter.c:
  audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
  ...and fix code style a bit.

2009-11-26 10:31:00 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/audio/gstaudiofilter.h:
  audiofilter: Add _CAST variants of the cast macros

2009-11-25 10:26:16 -0600  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  audiosink: add adjustement when slaving
  Our calibration against the pipeline clock is done with the adjusted
  ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
  when reusing audio sinks after switching clocks and slaving methods in a
  pipeline.

2009-11-25 16:17:13 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
  ffmpegcolorspace: Prefer transforming alpha formats to alpha formats and the other way around
  Fixes bug #602834 and #350748.

2009-11-25 00:46:55 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
  oggdemux: Reset last_granule during seeking
  Fix case where we would reconstruct the wrong granulepos for
  outgoing streams immediately after a seek.

2009-11-24 22:08:09 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
  oggdemux: Fix timestamp generation for theora
  Timestamp generation was broken by the last commit for formats
  with a non-zero granule shift.  Also keep track of the last keyframe
  so that we can regenerate granulepos for theora.

2009-11-24 21:22:03 -0800  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
* ext/ogg/vorbis_parse.c:
  oggdemux: Fix vorbis parsing
  Add a granule to granulepos conversion function.  Fix the duration
  function for vorbis.  Handle timestamps on header packets differently
  and be more careful about calculating OFFSET and OFFSET_END.  After
  this change, timestamps for vorbis don't exactly match up with the
  timestamps that vorbisparse outputs, but it's unclear if vorbisparse
  is actually correct and it would add a lot more code to make oggdemux
  match vorbisparse.  Fixes #602790.

2009-11-19 19:28:12 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Transform QoS events to be meaningful for upstream elements
  This is necessary because the sinks don't notice the group switches
  and the decoders/demuxers have a different running time than the
  sinks.
  Fixes bug #537050.

2009-11-21 22:05:34 +0100  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
  ogg: Fix generation of timestamps and durations
  After changing some internal functions, I forgot to update
  the code that puts the values on the buffers.

2009-08-29 10:51:48 -0700  David Schleef <ds@schleef.org>

* ext/ogg/Makefile.am:
* ext/ogg/dirac_parse.c:
* ext/ogg/dirac_parse.h:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstoggstream.c:
* ext/ogg/gstoggstream.h:
* ext/ogg/vorbis_parse.c:
  ogg: Add ogg stream parsing
  Adds code that parses headers of various formats encapsulated in
  Ogg in order to calculate timestamps and durations of each buffer.
  Removes the creation of helper decoder elements to do this calculation
  via conversion queries.
  Fixes: #344013, #568014.

2009-09-04 00:11:38 -0700  David Schleef <ds@schleef.org>

* ext/ogg/gstoggmux.c:
  oggmux: don't overwrite object properties

2009-11-21 17:54:49 +0200  Stefan Kost <ensonic@users.sf.net>

* ext/theora/theoradec.c:
  debug: also cast packet.packetno to gint64 in debug log
  We do this already for granulepos to handle ogg_int64_t mismatches.

2009-11-21 17:47:26 +0200  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/audio/gstbaseaudiosrc.c:
  debug: fix format string that was missing a var

2009-10-10 00:32:04 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/adder/gstadder.c:
* tests/check/elements/adder.c:
  adder: make events succeed, if they succed on atleast one pad

2009-11-19 14:51:33 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: error when all streams have no buffers
  In some cases (all buffers dropped by a parser) a decodebin2
  chain might receive an EOS before it gets enough data to
  expose a decoded pad. In the case that no streams can expose
  a pad we should error out instead of hang.
  Fixes #542758

2009-11-19 12:23:08 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Fix stupid bug introduced in last commit

2009-11-19 12:10:58 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Aggregate the stream-changed message by looking at the seqnum
  Just counting how many messages were sent and how many were received
  is not good enough because they might've been duplicated (e.g. by the
  visualization audio tee). Comparing the sequence numbers should give
  better results in that case.

2009-11-19 10:05:28 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Ignore async state changes of the uridecodebins
  Otherwise the async state change from READY->PAUSED of the
  uridecodebins will take playbin2 from PLAYING->PAUSED again
  during gapless group switches.
  Fixes bug #602000.

2009-11-19 10:30:06 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 0702fe1 to 53a2485

2009-11-18 14:50:28 -0300  Thiago Santos <thiago.sousa.santos@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: set to buffer less on no-more-pads
  When a decodebin2 receives no-more-pads of a group it
  can set that group's multiqueue buffering thresholds to
  'playing' buffering method, avoiding that it buffers
  too long and cause problems when using with queue2.
  See the associated bug for details.
  Fixes #600787

2009-11-18 17:09:28 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  baseaudiosink: fix initial calibration
  When we are calibrating the internal clock against the external clock take into
  account the time offset applied to our internal clock because we will subtract
  that in the render_function again.

2009-11-18 09:22:39 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Don't handle DURATION queries during group switches
  During a group switch return the cached duration of the old group
  because the old group still didn't finish playback. If we have no
  cached duration return FALSE.
  Fixes bug #585969.

2009-11-15 19:36:21 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Post a stream-changed message after activating a group
  This is useful to detect when playbin2 has really switched to the next
  group after about-to-finish for example.
  Fixes bug #584987.

2009-11-18 12:27:19 +0000  Jan Schmidt <thaytan@noraisin.net>

* win32/common/libgstvideo.def:
  win32: Add new still-frame API to the defs
  Add gst_video_event_new_still_frame() and
  gst_video_event_parse_still_frame() functions to the win32 defs files

2009-11-18 12:37:44 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosrc.c:
  baseaudiosrc: fix 'uninitialized' compiler warning

2009-11-18 10:14:41 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  configure: bump core requirement to 0.10.25.1
  We depend on new API that's only in git so far.

2009-11-15 17:34:37 +0000  Jan Schmidt <thaytan@noraisin.net>

* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
* tests/check/libs/video.c:
  video: Add functions to create/parse still frame events.
  Add a new video event to mark the start or end of a still-frame
  sequence, and a parser function to identify and extract info from
  such events.
  API: gst_video_event_new_still_frame()
  API: gst_video_event_parse_still_frame()
  Fixes: #601942

2009-11-17 16:39:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: make sure we always go to PAUSED async
  Set the need_async_start flag before going to PAUSED so that we always post the
  ASYNC_START message, even after reusing playsink.

2009-11-17 16:37:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: make sure we remain a sink
  When we remove our elements, we could lose our sink flag. Make sure we remain a
  sink by setting the flag again after removing elements.

2009-11-16 22:47:54 +0200  Stefan Kost <ensonic@users.sf.net>

* gst/audioconvert/gstaudioconvert.c:
  audioconvert: remove unused array

2009-11-16 09:57:56 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/subparse/gstsubparse.c:
  subparse: Use new double->fraction transformation function from core

2009-11-14 14:05:43 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Make subtitle error handling more robust and ignore late errors too
  Make sure, to only "simulate" subtitle no-more-pads if it was still
  pending and also handle errors in the subtitle pipeline as warnings
  after the subtitles prerolled.
  Don't set the suburidecodebin to READY after errors, handle_message
  will usually be called from the streaming thread and doing that
  from there is obviously not a good idea.

2009-11-14 13:21:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
  subtitleoverlay: Handle errors from subtitle elements as warning and go into passthrough mode

2009-11-13 12:47:55 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Don't leak the GError and debug string when parsing error messages

2009-11-13 11:16:44 +0100  Sreerenj B <bsreerenj@gmail.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: avoid crashing on SIGPIPE
  Use send() instead of write() so that we can pass the MSG_NOSIGNAL flags to
  avoid crashing with SIGPIPE when the remote end is not listening to us anymore.
  Fixes #601772

2009-11-11 17:35:45 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Improve subtitle passthrough in uridecodebin
  Now the caps property isn't set anymore for the subtitle caps
  but instead in the autoplug-continue signal it is detected
  if the caps belong to a supported subtitle stream.
  This makes automatic use of newly installed plugins.

2009-11-11 17:08:47 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
  subtitleoverlay: Only recreate factory caps if necessary and cache them

2009-11-10 18:27:15 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
  subtitleoverlay: Only update the factory list when the registry has changed
  Also don't free the list every time we go to NULL.

2009-11-08 15:04:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
  subtitleoverlay: Use gst_pad_get_caps_reffed()

2009-11-07 21:38:10 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
  playbin2/playsink: Use new "silent" property instead of unlinking
  This makes sure that subtitleoverlay still gets segment updates and
  everything to pass on downstream. Without this segment problems happen.

2009-11-07 21:10:27 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
  subtitleoverlay: Update segments after pushing the events downstream
  This makes sure that we don't apply segments twice downstream. Also
  always send our newsegment events downstream.

2009-11-07 21:09:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
  subtitleoverlay: Add silent property to disable subtitles
  This tries to disable subtitles in the overlay or renderer
  and if that's not possible it goes into passthrough mode.

2009-11-07 11:46:49 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
  subtitleoverlay: Set the video framerate on parsers if possible
  Fixes bug #599649.

2009-11-07 11:31:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
  subparse: Make fps a GstFraction typed property and use it properly

2009-11-07 11:08:19 +0100  Iago Toral <itoral@igalia.com>

* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
  subparse: Add property for the video framerate

2009-11-06 12:51:22 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Handle external subtitles better
  First of all, make sure that suburidecodebin never
  errors out because of not-linked in case external subtitles
  are used but then subtitles are disabled.
  And then make sure that external subtitles always start from
  the correct position and are not racing until EOS if they
  get unselected and selected again.

2009-11-04 17:29:07 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Flush the subtitles before switching to a new subtitle stream
  This makes sure that all currently shown subtitles disappear
  and new ones can be shown as soon as possible.

2009-11-03 12:47:55 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Set subtitle caps as raw caps for the uridecodebins
  This will make sure that no subparse is ever plugged and subtitleoverlay,
  that subpicture streams are handled the same was as subtitles and that
  subtitle renderers are used if available.
  Fixes bugs #595123, #570753, #591662, #591706.

2009-11-03 12:33:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
  playbin2/playsink: Remove everything related to subpicture streams
  These will soon be handled the same way as subtitle streams.

2009-11-02 15:50:17 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Add a queue before subtitleoverlay
  This will improve playback, and the same thing is done
  for subpicture streams too.

2009-11-02 15:05:41 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Use subtitleoverlay for subtitles

2009-11-02 07:43:42 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
  subtitleoverlay: Add to the docs

2009-10-13 16:48:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/Makefile.am:
* gst/playback/gstplayback.c:
* gst/playback/gstsubtitleoverlay.c:
* gst/playback/gstsubtitleoverlay.h:
  subtitleoverlay: Add new element for generic subtitle overlaying
  This autopluggs the required elements for parsing and rendering
  different subtitle formats on a video stream.
  Fixes bug #600370.

2009-11-11 19:32:01 -0500  Olivier Crête <olivier.crete@collabora.co.uk>

* ext/theora/theoradec.c:
  theoradec: Keep timestamp from incoming buffer if it is valid
  Fixes bug #601627.

2009-11-11 14:00:26 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
  playback: Update factories list on every access if the registry has changed
  This makes application's simpler because the element doesn't need to
  go to NULL first to make use of newly installed plugins.
  Fixes bug #601480.

2009-11-10 18:13:25 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
  playback: When going from NULL->READY check if the registry has new features
  This makes it possible to use newly installed plugins after going back
  to NULL instead of requiring a new instance.
  Fixes bug #599266.

2009-11-10 13:55:26 +0000  Jan Schmidt <thaytan@noraisin.net>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: Clear the EOS state on a seek.
  Allow seeking back into the stream after it hits EOS.

2009-11-10 12:21:50 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/audioresample/README:
* gst/audioresample/arch.h:
* gst/audioresample/fixed_arm4.h:
* gst/audioresample/fixed_arm5e.h:
* gst/audioresample/fixed_bfin.h:
* gst/audioresample/fixed_debug.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample_sse.h:
* gst/audioresample/speex_resampler.h:
  audioresample: Update speex resampler to latest GIT

2009-11-10 00:48:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: assign chain->mute before using it
  Fixes GObject warnings when starting totem.

2009-10-28 22:10:33 -0700  David Schleef <ds@schleef.org>

* ext/theora/theoradec.c:
  theora: Fix alignment of frames when converting
  Fix logic inversion in calculating the offset in the theora
  frame when copying to a GStreamer frame.

2009-11-09 19:58:20 +0100  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstfactorylists.c:
  playback: Fix the order in strcmp that I broke in previous commit.

2009-11-09 19:16:21 +0100  Edward Hervey <bilboed@bilboed.com>

* gst/typefind/gsttypefindfunctions.c:
  typefind: Reduce number of calls to gst_type_find_peek.
  Shaves off a couple percents off typefinding

2009-11-09 17:49:51 +0100  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstfactorylists.c:
  playback: Avoid expensive API calls in tight loop.
  We know we're dealing with GstPluginFeature.

2009-11-09 18:11:42 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/libs/cddabasesrc.c:
  cddabasesrc: Add unit test for property settings
  Also includes a regression test for bug #601104.

2009-11-09 18:04:23 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/cdda/gstcddabasesrc.c:
  cddabasesrc: Never return a negative track number in get_uri()

2009-11-09 18:03:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/cdda/gstcddabasesrc.c:
  cddabasesrc: Don't set the track to 1 every time a device is set
  Fixes bug #601104.

2009-11-08 11:27:10 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstinputselector.c:
  inputselector: Remove useless variables and fix a uninitialized variable compiler warnings

2009-11-06 17:01:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Add property to disable/enable posting of stream-topology messages
  Most people don't need this messages and generating them is quite
  expensive.

2009-11-06 15:12:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Protect subtitle elements and subtitle encoding by a new mutex
  Using the object lock here can and will lead to deadlocks because
  of deep-notifies of property changes: the deep-notify handler will
  get the parent of objects, which will take the object lock again.
  Fixes bug #600479.

2009-11-06 13:13:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstinputselector.c:
  inputselector: Make sure that running_time->timestamp calculation never becomes negative

2009-11-06 13:25:05 +0200  Mart Raudsepp <leio@gentoo.org>

* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
  examples: Correct casting of g_signal* funcs first arguments
  This completes the deprecated GTK API fix in commits 81a0a986 and
  79adfa54 - unlike gtk_signal_connect and co, g_signal_connect and
  co take a gpointer, not a GtkObject.

2009-11-06 12:25:53 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Improve all-raw-caps detection for pads

2009-11-06 12:19:47 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosrc.c:
  basesrc: fix startup position in the ringbuffer
  When we start and we need to produce the first sample, go to the next sample
  that will be written into the ringbuffer instead of trying to go to sample 0.
  We relied on rather small ringbuffer sizes to correctly go to the current
  sample, which breaks whith large buffers.
  Fixes #600945

2009-11-06 11:26:14 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstinputselector.c:
  inputselector: Use the start time (i.e. timestamp) as the last stop
  Using the end time makes it impossible to replace buffers, which is
  a big problem for subtitles that could have very long durations.

2009-11-06 12:08:19 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Synchronize video/text based on the running time
  Instead of simply using the buffer timestamps.

2009-11-06 09:30:38 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Clip text buffers to the text segment and reset segments properly

2009-11-06 09:01:34 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
  textoverlay: Put the video segment into the instance struct instead of allocating it separately

2009-11-06 09:05:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Check if text timestamp/duration is valid before clipping

2009-11-05 23:33:42 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/theora/theoradec.c:
  theoradec: printf format fix

2009-11-05 15:42:09 +0100  Olivier Crête <olivier.crete@collabora.co.uk>

* gst/gdp/gstgdpdepay.c:
  gdpdepay: Clear adapter on flush and state change
  Fixes #600469

2009-11-05 13:12:19 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstinputselector.c:
  inputselector: use _get_caps_reffed()

2009-11-05 13:00:27 +0200  Stefan Kost <ensonic@users.sf.net>

* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
  pad: rename new api from _refed to _reffed.
  Due to popular demand rename the new api as we still can.

2009-11-04 18:57:07 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
  playbin2: avoid copying caps
  Use get_caps_refed() when we can.

2009-11-04 18:31:09 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: use new getcaps function to avoid copies
  Use the gst_pad_get_caps_refed() to avoid some caps copy functions.

2009-11-04 17:50:11 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: use faster element_link_pads
  Use the faster gst_element_link_pads because we know for sure the sinkpad name
  and we don't need to have the function search for a suitable pad anymore.

2009-11-04 16:16:31 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  baseaudiosink: make drift tolerance configurable
  Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
  drift or timestamp drift instead of relying on the latency-time value for clock
  drift and 500ms for timestamp drift.
  Remove warning about discont timestamp and simply resync. The warning is in some
  cases not correct and is triggered more frequently now that we lower the
  tolerance value.

2009-11-04 10:52:06 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Return NOT_LINKED for unselected text pads from a demuxer
  We want to return NOT_LINKED for unselected pads but only for pads
  from the normal uridecodebin. This makes sure that subtitle streams
  are not raced past audio/video from decodebin2's multiqueue.
  For pads from suburidecodebin OK should always be returned, otherwise
  it will most likely stop with an error.

2009-11-04 08:20:59 +0100  Stefan Kost <ensonic@users.sf.net>

* gst/playback/gstinputselector.c:
  inputselector: also add inline to the proto to fix the build
  Merged from gst-plugins-bad, e1e9be6dbe1bd0df0543f2a72dcf9cc6d644dd78.

2009-11-03 12:01:16 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Initialize caps property with the default raw caps

2009-11-03 11:48:40 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/Makefile.am:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstrawcaps.h:
  decodebin2: Use static caps for the default raw caps and put them into a separate header
  This way we can use the same default raw caps everywhere.

2009-11-03 08:26:37 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: First draw outline text and then the real text
  Improves the output a bit because no parts of the outline are
  overwritten again.

2009-10-31 14:02:40 +0100  Josep Torra Valles <n770galaxy@gmail.com>

* gst/playback/gstplaybin.c:
  playbin: Make sure to keep a reference on the volume element
  Fixes null pointer dereferences under certain circumstances.
  Fixes bug #595401.

2009-10-31 09:47:54 +0100  Edward Hervey <bilboed@bilboed.com>

* po/POTFILES.in:
  po: queue2 has moved to core

2009-10-30 09:24:30 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Reset {mute,volume}-changed flags after setting the volume
  These flags are there to make sure that the volume is set, if there
  is no volume element yet.

2009-10-30 09:24:03 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: If notify::{volume,mute} is triggered by the volume element, update our internal state

2009-10-29 14:30:31 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Proxy notify::volume and notify::mute from the volume/mute elements (or sinks)
  Fixes bug #600027.

2009-10-29 14:19:09 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Proxy notify::volume and notify::mute from the playsink to playbin2

2009-10-29 11:37:04 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* docs/plugins/inspect/plugin-queue2.xml:
  queue2: Remove inspect file

2009-10-29 11:29:46 +0100  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c:
  queue2: Remove from gst-plugins-base
  This is now in coreplugins.

2009-10-28 11:29:36 +0200  Stefan Kost <ensonic@users.sf.net>

* docs/libs/gst-plugins-base-libs-docs.sgml:
  docs: include more indexes

2009-10-28 11:13:20 +0200  Stefan Kost <ensonic@users.sf.net>

* docs/libs/gst-plugins-base-libs-docs.sgml:
  docs: turn entities into xi:includes
  This is faster to process and easier to maintain. Its also less 80s.

2009-10-28 10:17:43 +0200  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/rtp/gstrtpbuffer.c:
  rtp: dump packets which we reject

2009-10-28 01:01:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/check/pipelines/.gitignore:
  .gitignore: ignore basetime unit test binary

2009-10-28 00:59:35 +0000  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstinputselector.c:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstqueue2.c:
* gst/playback/gststreaminfo.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstssaparse.c:
  Remove GST_DEBUG_FUNCPTR where they're pointless
  There's not much point in using GST_DEBUG_FUNCPTR with GObject
  virtual functions such as get_property, set_propery, finalize and
  dispose, since they'll never be used by anyone anyway. Saves a
  few bytes and possibly a sixteenth of a polar bear.

2009-10-27 15:23:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstqueue2.c:
  queue2: add custom acceptcaps function

2009-10-27 15:22:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: implement low/high watermark property

2009-10-23 14:56:11 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* tests/examples/seek/seek.c:
  seek: add checkbox to enable buffering

2009-10-23 14:54:47 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: don't use 2 buffering elements
  Only use the multiqueue buffering when we don't have a stream (and thus are
  using queue2 to do the buffering already).

2009-10-23 14:34:42 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplay-enum.c:
* gst/playback/gstplay-enum.h:
* gst/playback/gstplaybin2.c:
  playbin2: add flag to enable decodebin buffering
  Add a flag that enables buffering in decodebin.

2009-10-23 14:32:29 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: buffering is implemented now

2009-10-23 14:30:52 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: buffering is implemented now

2009-10-23 14:09:17 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: configure use-buffering on multiqueue

2009-10-23 13:58:25 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: use 0 for max buffer size

2009-10-23 13:53:21 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: set some reasonable defaults

2009-10-23 13:44:12 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: set buffering properties on decodebin2
  Propagate the buffering properties on decodebin2 but only if we are not already
  doing download buffering.

2009-10-23 11:52:09 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: add use-buffering property
  Add a use-buffering property that will perform buffering on the parsed or
  demuxed media.

2009-10-23 11:31:47 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: refactor queue size configuration.
  Refactor the queue size configuration into a new method.
  Use the same queue values for buffering as for preroll.

2009-10-23 11:08:50 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: move error path down

2009-10-23 11:02:40 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: implement max queue size properties

2009-10-23 10:42:23 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: add properties for buffering
  Add properties that can be used to configure the multiqueue buffers and
  buffering methods

2009-10-24 13:19:08 +0200  Edward Hervey <bilboed@bilboed.com>

* tests/examples/app/Makefile.am:
* tests/examples/seek/Makefile.am:
* tests/examples/v4l/Makefile.am:
  examples: fix linking order.
  the uninstalled wrapper would create a LD_LIBRARY_PATH with system-wide
  path before the local ones... resulting in the example applications picking
  up the system-wide libraries and not the (potentially modified) uninstalled
  libraries

2009-10-24 13:08:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Don't destroy the suburidecodebin on errors
  It can still be reused

2009-10-24 13:07:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: If setting the state of the suburidecodebin fails just warn, don't error out

2009-10-24 12:12:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Don't set uridecodebin states to NULL before reusing them
  This makes sure that the internal decodebin2 and everything else can
  be reused without reinstantiation.

2009-10-18 17:28:22 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Store unused decodebin2 instances for further usage.
  This allows faster re-use of uridecodebin.
  https://bugzilla.gnome.org/show_bug.cgi?id=599471

2009-10-23 17:49:15 -0700  David Schleef <ds@schleef.org>

* ext/theora/gsttheoraparse.h:
* ext/theora/theoraparse.c:
  theora: Convert theoraparse to libtheora 1.0 API

2009-10-21 12:38:59 +0300  Olivier Crête <olivier.crete@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  rtpaudiopayload: Only sent exact multiple of the frame size
  Also align the maximum size with the frame size, not only the minimum

2009-10-22 09:12:03 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

* gst/audiorate/gstaudiorate.c:
  audiorate: move debug calculation into debug macro
  Remove in_duration and move its calculation to
  GST_LOG_OBJECT macro. This way it will only be calculated
  if we have debug enabled.

2009-10-22 09:06:02 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

* gst/audiorate/gstaudiorate.c:
  audiorate: Removing unused variable
  The in_stop variable was never read. Removing it.

2009-10-22 08:40:01 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>

* gst/audiorate/gstaudiorate.c:
  audiorate: be more accurate on offset math
  Replace gst_util_uint64_scale_int for its rounding version
  to improve accuracy and avoid inserting samples where
  they aren't needed.
  Fixes #499181

2009-10-22 10:17:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Optimize a bit more
  ...and add a FIXME for bug #598695 and explain
  what we should do once Pango supports user fonts.

2009-10-22 10:02:11 +0200  Iago Toral <itoral@igalia.com>

* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c:
  subparse: Add support for DKS subtitle format
  Fixes bug #598936.

2009-10-22 09:31:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Do shading as first operation

2009-10-22 09:08:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: Only use a single cairo surface for drawing
  ... and comment/optimize what is going on here a bit better.

2009-10-21 16:24:29 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstinputselector.c:
  inputselector: set output caps before pushing
  Set the output caps on the srcpad before pushing the buffer because else core
  will do a rather expensive check to see if we can actually accept those caps on
  the srcpad.

2009-10-21 15:58:11 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstinputselector.c:
  inputselector: install an acceptcaps function
  Install a custom acceptcaps function instead of using the default expensive
  check. We accept whatever downstream accepts so we pass along the acceptcaps
  call to the downstream peer.

2009-10-21 20:35:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefind: fix typo in previous mxf typefinder change

2009-10-21 20:44:33 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/typefind/gsttypefindfunctions.c:
  typefind: speed up mxf_type_find over 300 times for worst case scenarios
  * memcmp is expensive and was being abused, reduce calling it by checking
  the first byte.
  * iterating one byte at at time over 64 kbites introduces a certain overhead,
  therefore we now do it in chunks of 1024 bytes
  And I do mean over 300 times. The average instruction call per mxf_type_find
  was previously 785685 and it's now down to 2458 :)

2009-10-20 17:13:39 -0400  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstfactorylists.c:
  decodebin2: avoid type checks

2009-10-20 09:00:28 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstdecodebin2.c:
  gst/decodebin2: Ensure we get fixed caps for topology message
  There are some corner cases (like with dvdemux amongst others) where
  the caps won't be negotiated, but the pad has fixed caps.

2009-10-20 08:52:36 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstdecodebin2.c:
  gst/decodebin2: Don't expose chains if we're shutting down.
  This avoids adding flushing pads to ourself

2009-10-17 21:16:57 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* ext/pango/gsttextoverlay.c:
  pango: bump pango requirement to stable version and remove ifdefs
  Bump pango requirement from an ancient development version to an
  ancient stable version.

2009-10-17 21:11:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/rtsp/.gitignore:
  .gitignore: update after files got renamed

2009-10-16 10:54:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  basertppayload: small comment fix

2009-10-16 10:50:35 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtp/gstbasertppayload.c:
  rtp: Correct timestamping of buffers when buffer_lists are used
  The timestamping of buffers when buffer_lists are used failed if
  a buffer did not have both a timestamp and an offset.

2009-10-16 10:56:56 +0300  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp-marshal.list:
* gst-libs/gst/rtsp/gstrtspextension.c:
* gst-libs/gst/rtsp/rtsp-marshal.list:
* gst-libs/gst/video/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
  build: fix previous commit to fully accomodate the glib-gen.mak changes
  I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
  marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.

2009-10-16 10:18:45 +0300  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
  build: use gst-glib-gen.mak to fix the glib build rules. Fixes #598114
  The build rules in glib-gen.mak were using pattern rules in a non save way.

2009-10-16 10:14:36 +0300  Stefan Kost <ensonic@users.sf.net>

* common:
  Automatic update of common submodule
  From 85d1530 to 0702fe1

2009-09-10 11:39:18 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoradec.c:
  theora: Make theoradec use gstvideo for image conversion
  Vastly simplifies code.
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-10 09:36:31 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoradec.c:
  theora: Don't always round to even width/height
  Previously, the code always rounded to even sizes. Now it only ensures
  that pic_x and pic_y are multiples of 2 if the output format requires
  it.
  Also inlcudes fixes to take pic_x/y into account properly when copying
  the buffer.
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-10 00:00:44 +0200  Benjamin Otte <otte@gnome.org>

* configure.ac:
  theora: Don't check for theora.pc anymore
  THe new APIs from theoradec and theoraenc are used now.
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

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
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-09 23:44:36 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/Makefile.am:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Port encoder to new Theora API
  Includes ripping out the old buffer copy code to fill up to frame size.
  This is not necesary with the new encoder.
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-09 21:59:31 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Disable sharpness property
  It's ignored by libtheora
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-09 21:57:08 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Disable noise-sensitivity property
  It is ignored by libtheora
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-09 21:50:57 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Disable keyframe-mindistance property
  It's ignored by the current Theora library
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-09 21:48:08 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Disable keyframe_threshold property
  It's ignored by the current theora encoder
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-09 20:26:47 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Get rid of "quick" property
  The proeprty is not used by libtheora at all
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-08 15:12:23 +0200  Benjamin Otte <otte@gnome.org>

* configure.ac:
* ext/theora/theoraenc.c:
  theora: remove support for outdated granulepos hack
  This is in preparation to switching to switching to the new Theora API
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-08 13:23:04 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Ignore border property
  Always make the video use black as padding color.
  The output will be identical to previous versions.
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-09-08 13:18:26 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c:
  theora: Ignore the center property, always set video to top left
  This is not a necessary property, the output will be identical no matter
  what.
  https://bugzilla.gnome.org/show_bug.cgi?id=594729

2009-10-15 16:34:28 +0100  Jan Schmidt <thaytan@noraisin.net>

* po/Makevars:
  po: Don't create backup .po files
  As well as preventing creation of useless backup files, it works
  around a bug in gettext 0.17 on OS/X

2009-10-15 13:13:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Post a element message on the bus with the stream topology
  Fixes bug #598533.

2009-10-15 13:01:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Store the "endcaps" of a chain
  This are the caps that either resulted in a deadend if
  no plugin for them could be found or raw caps.

2009-10-15 11:38:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Store for every chain, which pad resulted in its creation

2009-10-15 10:28:39 +0100  Jan Schmidt <thaytan@noraisin.net>

* tests/check/pipelines/basetime.c:
  check: Don't fail the basetime test when no audiosrc is available
  On OS/X the DEFAULT_AUDIOSRC is not going to be available, because
  it isn't in gst-plugins-base. Just defer the test, instead of
  failing it.

2009-10-14 10:41:03 +0200  Edward Hervey <bilboed@bilboed.com>

* common:
  Automatic update of common submodule
  From a3e3ce4 to 85d1530

2009-10-14 08:36:54 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Use gst_object_has_ancestor() instead of our own implementation of it

2009-10-13 19:14:41 +0300  Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>

* gst-libs/gst/audio/gstbaseaudiosrc.c:
  baseaudiosrc: fix timestamp comparission, Fixes #597407

2009-10-13 13:52:02 +0300  Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>

* tests/check/Makefile.am:
* tests/check/pipelines/basetime.c:
  tests: new test for baseaudiosrc base_time comparison
  This test reveals a bug in comparison operation between timestamp and
  GstElement's base_time in GstBaseAudioSrc.

2009-10-08 19:55:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Don't stop completely on initialization errors from subtitle elements
  Instead disable the subtitles and play the other parts of the stream.
  Fixes bug #587704.

2009-10-13 16:50:37 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Ignore no-more-pads from non-demuxer elements
  instead of printing an error that no corresponding group could
  be found. no-more-pads from non-demuxer elements doesn't give
  any additional information because there can only be a single srcpad.
  Fixes bug #598288.

2009-10-12 21:30:15 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/audioconvert/gstaudioconvert.c:
  audioconvert: track active conversion in perf log

2009-10-12 15:48:46 +0200  Patrick Radizi <patrick.radizi at axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: handle socket errors
  gstrtspconnection.c:gst_rtsp_connection_receive() can hang when an error occured
  on a socekt. Fix this problem by checking for error on 'other' socket after poll
  return.
  Fixes #596159

2009-10-06 14:08:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstaudioclock.c:
  audioclock: whitespace fixes

2009-10-06 14:07:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/theora/theoradec.c:
  theoradec: avoid confusing error

2009-10-09 22:00:45 +0200  Josep Torra <n770galaxy@gmail.com>

* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
  vorbis: fixes warings in macosx snow leopard

2009-10-09 18:52:12 +0200  Josep Torra <n770galaxy@gmail.com>

* ext/theora/theoradec.c:
* ext/theora/theoraparse.c:
  theora: fixes warnings on macosx snow leopard

2009-10-09 16:56:29 +0200  Josep Torra <n770galaxy@gmail.com>

* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
  ogg: fixes warnings on macosx snow leopard

2009-10-09 16:19:17 +0200  Josep Torra <n770galaxy@gmail.com>

* ext/ogg/gstoggdemux.c:
  oggdemux: fix a warning in macosx

2009-10-08 14:16:44 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst-libs/gst/tag/tags.c:
  tag: use BOM to recognize UTF-16/32 encoding and convert accordingly

2009-10-09 15:11:16 +0100  Jan Schmidt <thaytan@noraisin.net>

* tests/check/gst-plugins-base.supp:
  check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty.

2009-10-09 15:32:45 +0200  Josep Torra <n770galaxy@gmail.com>

* ext/gnomevfs/gstgnomevfssrc.c:
  audioconvert: change the format instead of cast as ensonic asked

2009-10-09 15:29:15 +0200  Josep Torra <n770galaxy@gmail.com>

* gst/audioconvert/gstchannelmix.c:
  audioconvert: fixes warning: format not a string literal and no format arguments
  redo of valid part of my previous revert.

2009-10-09 15:19:42 +0200  Josep Torra <n770galaxy@gmail.com>

* common:
* gst/audioconvert/gstchannelmix.c:
  Revert "audioconvert: fixes warning: format not a string literal and no format arguments"
  Revert this commit as unintentionally I've changed common.
  This reverts commit 49ea0138223ec5f9e53780635cbcc70f33778667.

2009-10-09 14:28:42 +0200  Josep Torra <n770galaxy@gmail.com>

* ext/gnomevfs/gstgnomevfssrc.c:
  gnomevfssrc: fixes warnings in macosx
  warning: format '%llu' expects type 'long long unsigned int', but argument 8 has type 'GnomeVFSFileOffset'
  warning: format '%lld' expects type 'long long int', but argument 9 has type 'guint64'

2009-10-09 14:23:36 +0200  Josep Torra <n770galaxy@gmail.com>

* gst/videorate/gstvideorate.c:
  videorate: fix warning in macosx

2009-10-09 14:20:47 +0200  Josep Torra <n770galaxy@gmail.com>

* gst/audiorate/gstaudiorate.c:
  audiorate: fix warning in macosx

2009-10-09 14:14:15 +0200  Josep Torra <n770galaxy@gmail.com>

* common:
* gst/audioconvert/gstchannelmix.c:
  audioconvert: fixes warning: format not a string literal and no format arguments

2009-10-09 14:07:24 +0200  Josep Torra <n770galaxy@gmail.com>

* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
  audio: fix warnings building on macosx

2009-10-08 18:08:22 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
  ffmpegcolorspace: chwck formats just once per _chain()

2009-10-08 17:49:39 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
  ffmpegcolorspace: add perf-log-category and log suboptimal operation
  Log if we use an intermediate colorspace for conversion.

2009-10-08 10:59:36 +0100  Jan Schmidt <thaytan@noraisin.net>

* common:
  Automatic update of common submodule
  From 19fa4f3 to a3e3ce4

2009-10-08 00:17:21 +0100  Jan Schmidt <jan.schmidt@sun.com>

* gst/playback/gstdecodebin2.c:
  decodebin2: Fix type-punning warning

2009-09-26 12:56:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Chains with an exposed endpad are complete too
  This allows partial group changes, i.e. demuxer2 in the example below
  goes EOS but has a next group and audio2 stays the same.
  /-- >demuxer2---->video
  demuxer---             \--->audio1
  \--->audio2

2009-09-26 12:47:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Use the iterate internal links function instead of string magic to get multiqueue srcpads

2009-09-24 14:56:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Don't post missing plugin messages twice
  decodebin2 already posts them after emitting the unknown-type signal,
  there's no need to post another one.

2009-09-26 12:17:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Rewrite autoplugging and how groups of pads are exposed
  This now keeps track of everything that is going on, creates
  a tree of chains and groups to allow "demuxer after demuxer" scenarios
  and allows chained Oggs with multiple streams (needs oggdemux or playbin2 fixes).
  Also document everything in detail and give a general overview of what
  decodebin2 is doing at the top of the sources.
  Fixes bug #596183, #563828 and #591677.

2009-10-07 17:45:33 +0300  Stefan Kost <ensonic@users.sf.net>

* sys/ximage/ximagesink.c:
  ximagesink: only start event thread if needed
  The event thread is doing 20 wakeups per second to poll the events. If one
  runs ximagesink with handle-events=false and handle-expose=false then we can
  avoid the extra thread.

2009-10-07 16:56:28 +0200  Edward Hervey <bilboed@bilboed.com>

* ext/theora/theoraenc.c:
  theoraenc: Make the default quality property 48.
  This guarantees that people who use theoraenc without modifying any
  properties will end up with a reasonably good quality output.
  48 is also the default of the encoder_example application shipped with
  libtheora.

2009-10-07 11:48:37 +0200  Benjamin Otte <otte@gnome.org>

* tests/check/libs/video.c:
  tests/check/libs/video.c: Update strides for Y41B

2009-10-07 10:32:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtspconnection: we can use GLib 2.18 API unconditionally now

2009-10-07 10:13:59 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
  configure: bump GLib requirement to 2.18
  Bump required GLib version as per the release planning docs.

2009-10-05 00:33:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/interfaces/tuner.c:
  docs: clarify GstTuner docs in two places

2009-09-25 15:32:18 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* sys/v4l/gstv4lelement.c:
  v4l: fix compiler warning
  Fix 'variable may be used uninitialized' compiler warning (which is
  true in theory, but can't actually ever happen, since we always
  call the function with check=FALSE).
  Fixes #596313.

2009-10-07 11:56:35 +0300  Stefan Kost <ensonic@users.sf.net>

* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstogmparse.c:
* gst/subparse/gstsubparse.c:
* gst/subparse/mpl2parse.c:
* gst/subparse/tmplayerparse.c:
  build: sprintf, sscanf need stdio.h

2009-09-15 15:26:06 +0300  Stefan Kost <ensonic@users.sf.net>

* sys/xvimage/xvimagesink.c:
  xvimagesink: only start event thread if needed
  The event thread is doing 20 wakeups per second to poll the events. If one runs
  xvimagesink with handle-events=false and handle-expose=false then we can avoid
  the extra thread.

2009-10-07 09:58:27 +0200  Benjamin Otte <otte@gnome.org>

* gst-libs/gst/video/video.h:
  Update Since tags for NV12/NV21
  They are added in 0.10.26 now, not 0.10.25

2009-09-23 15:31:50 +0200  Benjamin Otte <otte@gnome.org>

* gst/videotestsrc/videotestsrc.c:
  [videotestsrc] Make checkers-8 pattern create 8x8 instead of 16x16 tiles

2009-09-23 11:03:57 +0200  Benjamin Otte <otte@gnome.org>

* gst/ffmpegcolorspace/imgconvert_template.h:
  [ffmpegcolorspace] Fix NV12 and NV21 with odd width and height

2009-09-23 10:25:02 +0200  Benjamin Otte <otte@gnome.org>

* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
  Add NV12 and NV21 formats

2009-09-21 18:49:42 +0200  Benjamin Otte <otte@gnome.org>

* gst-libs/gst/video/video.c:
  [video] Fix Y41B
  Chroma components should be aligned on 4byte boundaries.
  https://bugzilla.gnome.org/show_bug.cgi?id=595849

2009-09-21 18:49:06 +0200  Benjamin Otte <otte@gnome.org>

* gst/videotestsrc/videotestsrc.c:
  [videotestsrc] Fix Y41B
  Chroma components should be aligned on 4byte boundaries.
  https://bugzilla.gnome.org/show_bug.cgi?id=595849

2009-10-07 07:28:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* configure.ac:
* gst-libs/gst/interfaces/streamvolume.c:
  streamvolume: Define cbrt() if it's not available
  Fixes build on Win32, bug #597537.

2009-09-24 16:05:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstfactorylists.c:
  factorylist: Use gst_caps_can_intersect() instead of _intersect()
  This is faster and results in less allocations.

2009-09-26 12:10:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Don't set the external ghostpads blocked but only their targets
  Pad blocks should never be done on external pads as outside elements
  might want to use their own pad blocks on them and this will lead to
  conflicts and deadlocks.

2009-09-26 12:04:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Only use the object lock for protecting the subtitle elements
  Using the decodebin lock will result in deadlocks if the subtitle encoding
  is accessed from a pad-added handler.

2009-09-26 18:11:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Improve debugging of pad blocks

2009-09-23 16:07:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
  playbin2/playsink: Use gst_object_ref_sink() instead of calling both separately

2009-10-06 19:59:11 -0700  David Schleef <ds@schleef.org>

* configure.ac:
  configure: Add an 'else' to pangocairo check
  Otherwise it exits if it fails.

2009-10-06 19:35:50 -0700  David Schleef <ds@schleef.org>

* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
  videotestsrc: add pattern with out-of-gamut colors
  Adds a pattern with out-of-gamut colors in a checkerboard
  pattern with in-gamut neighbors.  Useful for checking YCbCr->RGB
  color matrixing.  Correct matrixing and clamping will cause the
  checkerboard pattern to be invisible.

2009-10-06 19:17:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: use CLOSE_SOCKET() instead of close()
  Use CLOSE_SOCKET instead of directly calling close() because it does the right
  thing for windows.
  Fixes #597539

2009-10-01 14:19:41 +0200  Robert Swain <robert swain gmail com>

* gst/audioresample/gstaudioresample.c:
  audioresample: fix printf variable type
  Change printf variable type from %lu to %" G_GUINT64_FORMAT " as it
  should be for guint64.
  Fixes #596981

2009-09-30 23:22:35 +0100  Jan Schmidt <thaytan@noraisin.net>

* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
  ffmpegcolorspace: Use the ffmpegcolorspace debug category
  Move gstffmpegcodecmap debug to the ffmpegcolorspace category

2009-09-22 11:58:26 +0100  Jan Schmidt <thaytan@noraisin.net>

* gst/gdp/gstgdppay.c:
  gdppay: Don't repeat tags buffers for every new segment
  Only send a tag buffer when one is received, not after every new segment
  event/update.

2009-09-28 20:25:35 -0700  David Schleef <ds@schleef.org>

* gst/typefind/gsttypefindfunctions.c:
  typefind: detect 'ftypqt  ' as video/quicktime

2009-10-06 19:47:00 +0100  Jan Schmidt <thaytan@noraisin.net>

* configure.ac:
  back to development -> 0.10.25.1

=== release 0.10.25 ===

2009-10-05 13:56:15 +0100  Jan Schmidt <thaytan@noraisin.net>

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
  Release 0.10.25

2009-10-05 13:49:10 +0100  Jan Schmidt <thaytan@noraisin.net>

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

2009-10-01 17:17:55 +0100  Jan Schmidt <thaytan@noraisin.net>

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

2009-10-01 10:37:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
  pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB

2009-09-28 22:06:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: make the lock recursive for now
  Fixes #583255

2009-09-28 21:54:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: fix the vis property getter

2009-09-30 18:06:56 +0100  Christian F.K. Schaller <christian.schaller@collabora.co.uk>

* gst-plugins-base.spec.in:
  Add missing file to spec file

2009-09-17 16:57:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/cdda/gstcddabasesrc.c:
* tests/check/libs/cddabasesrc.c:
  cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc

2009-09-17 23:42:52 +1000  Jonathan Matthew <jonathan@d14n.org>

* gst-libs/gst/cdda/gstcddabasesrc.c:
* tests/check/libs/cddabasesrc.c:
  cddabasesrc: ignore URI fragments that look like device paths
  Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
  worked before the fix for bug #321532.
  Also adds a check for negative track numbers and some unit tests for URI
  parsing.
  Fixes bug #595454.

2009-09-17 01:20:45 +0100  Jan Schmidt <thaytan@noraisin.net>

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

2009-09-15 15:23:49 -0700  Michael Smith <msmith@songbirdnest.com>

* gst-libs/gst/tag/gstvorbistag.c:
  vorbistag: don't ever return NULL in list of strings.

2009-09-14 12:18:33 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstplaysink.c:
  playsink: Expose mute,volume,vis-plugin and font-desc properties
  https://bugzilla.gnome.org/show_bug.cgi?id=594623

2009-09-09 12:42:04 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstplaysink.c:
  GstPlaySink: Expose 'reconfigure' as an action signal.

2009-09-09 11:17:28 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstplaysink.c:
  GstPlaySink: Expose flags as a gobject property.

2009-09-08 11:35:20 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/playback/gstplayback.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
  playback: Register playsink as an element.
  This allows using playsink from outside the playback plugin.
  Add code to be able to request the sink pads using standard GStreamer API.
  TODO : expose GObject properties/signals.

2009-09-12 14:55:06 +0300  Stefan Kost <ensonic@users.sf.net>

* docs/libs/gst-plugins-base-libs.types:
  docs: add new gst_stream_volume_get_type to types file
  This is needs to get Gobject features to show up in the docs.

2009-09-12 15:48:11 -0700  David Schleef <ds@schleef.org>

* ext/ogg/gstoggdemux.c:
  oggdemux: Fix duration calculation for truncated files
  If the last page of a stream has a granulepos of -1, that is,
  it doesn't complete a packet, we need to continue to search
  for the last granulepos.

2009-09-12 14:01:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
  This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.

2009-09-12 02:23:07 +0100  Jan Schmidt <thaytan@noraisin.net>

* ext/theora/theoraenc.c:
  theoraenc: Fix a string leak in _getcaps()

2009-09-11 23:49:11 +0100  Jan Schmidt <thaytan@noraisin.net>

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

2009-09-11 21:44:18 +0100  Jan Schmidt <thaytan@noraisin.net>

* tests/check/elements/audioresample.c:
  check: Improve audioresample test
  Make the audioresample test work with CK_FORK=no, and
  turn a g_print into a GST_INFO.

2009-09-11 22:09:06 +0200  Benjamin Otte <otte@gnome.org>

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: Fix crashes with even widths
  The fix for green lines introduced by commit
  35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
  for even widths. This patch fixes it.

2009-09-11 15:11:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: Implement GstStreamVolume interface

2009-09-11 15:04:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
* tests/check/Makefile.am:
* tests/check/elements/volume.c:
  volume: Implement GstStreamVolume interface

2009-09-11 14:54:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/streamvolume.c:
* gst-libs/gst/interfaces/streamvolume.h:
* gst/playback/Makefile.am:
* win32/common/libgstinterfaces.def:
  interfaces: API: Add GstStreamVolume interface
  Fixes bug #567660.

2009-09-11 12:20:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: properly fix the HTTP manual mode
  When we're not parsing HTTP, return EPARSE when we get an HTTP
  message.

2009-09-11 10:16:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/interfaces/mixertrack.h:
  mixertrack: add READONLY and WRITEONLY flags
  Should really have been READABLE and WRITABLE, but those are hard to
  add whilst maintaining backwards compatibility. See #343615.
  API: GST_MIXER_TRACK_READONLY
  API: GST_MIXER_TRACK_WRITEONLY

2009-09-11 10:02:54 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/audio/gstringbuffer.c:
  ringbuffer: fix build against core that has debugging disabled
  The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.

2009-09-11 07:38:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/videorate/gstvideorate.c:
  videorate: Add Since marker for the new skip-to-first property

2009-09-11 07:36:10 +0200  Olivier Crête <olivier.crete@collabora.co.uk>

* gst/videorate/gstvideorate.c:
* gst/videorate/gstvideorate.h:
  videorate: Make videorate work with a live source
  Add a property that makes videorate skip to the first buffer it
  receives instead of padding the stream from segment start to the
  first real buffer.
  Fixes bug #567928.

2009-09-11 07:20:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/fft/gstfft.h:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.h:
  fft: Mark one function as const and add notes that the structs should be private in 0.11

2009-09-10 22:28:19 +0300  Stefan Kost <ensonic@users.sf.net>

* gst-libs/gst/audio/gstringbuffer.c:
  ringbuffer: add human readable format names when logging
  Add string array with human readable names for format and type to be used in log
  statements.

2009-09-10 18:19:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  basertppay: don't print RTP timestamps as clocktime
  Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
  Fixes #594757

2009-09-10 16:55:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
  playbin(2): Document that the volume property uses a linear scale
  Fixes bug #571610.

2009-09-10 14:04:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: don't return EPARSE
  Don't blindly return EPARSE when http mode is disabled.
  Restore old http mode after temporarily setting it to TRUE.

2009-09-10 12:38:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  baseaudiosink: add ugly backward compat hack
  Check for pulsesink < 0.10.17 because it includes code that is now included in
  baseaudiosink. Disable that code in baseaudiosink to be compatible with the
  older version.

2009-09-10 10:56:29 +0200  Benjamin Otte <otte@gnome.org>

* gst/ffmpegcolorspace/imgconvert.c:
  ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
  A green border could be visible when converting to Y444 or RGB, because
  the last chroma samples weren't copied correctly

2009-09-10 10:43:37 +0200  Benjamin Otte <otte@gnome.org>

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: Fix YVU9 and YUV9
  - Buffer sizes were computed different from ffmpegcolorspace
  - Green bar on right size for widths not divisable by 4

2009-09-10 10:08:28 +0200  Benjamin Otte <otte@gnome.org>

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: Fix image for odd widths in some formats
  videotestsrc rounds chroma down. This causes it to omit the last chroma
  value completely for odd widths when the chroma is downsampled.
  This patch special cases the last pixel to not be rounded down.

2009-09-10 10:02:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
  oggdemux: Handle kate and cmml as sparse streams too

2009-09-10 10:00:16 +0200  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
  oggdemux: Better handling of sparse streams by sending segment updates
  Fixes bug #397419.

2009-09-10 09:43:28 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/playback/gsturidecodebin.c:
  docs: tell a biit more about uri-decodebin and buffering

2009-09-09 18:24:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  baseaudiosink: take clock time in setcaps
  Take the time of the clock so that the last_time field is set. This is important
  for sinks that restart their internal ringbuffer after a caps change and need to
  know the last know position.

2009-09-09 18:24:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstaudioclock.c:
  audioclock: add some more debug

2009-09-09 16:44:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/theora/theoraenc.c:
  theoraenc: Print a debug message with supported formats

2009-09-07 17:29:38 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoraenc.c:
  theora: Check supported input formats in getcaps function
  We want to fail early when an older libtheora release is used that does
  not support Y444 or Y42B formats, so use a getcaps function that does
  this.

2009-09-04 21:37:04 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoraenc.c:
  theora: Implement support in theoraenc for Y444 and Y42B
  Fixes bug #594165.

2009-09-04 20:23:52 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoraenc.c:
  theora: Refactor the buffer copy code

2009-09-04 16:59:49 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoraenc.c:
  theora: Split yuv_buffer creation into its own function

2009-09-04 16:49:08 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoraenc.c:
  theora: Split out buffer resize in its own function

2009-09-04 14:06:09 +0200  Benjamin Otte <otte@gnome.org>

* ext/theora/theoraenc.c:
  theora: Add assertions that functions don't fail
  Some functions in libtheora can return an error, but that error cannot
  ever happen inside theoraenc. In those cases assert that it doesn't.

2009-09-09 16:21:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* tests/examples/seek/seek.c:
  seek: make stop state configurable
  Make it easy to experiment with different stop states (NULL and READY)

2009-09-09 16:19:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.c:
  baseaudiosink: correct for clock reset
  When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
  also make sure that the clock is updated with the elapsed time so that it
  alsways increments even when the ringbuffer goes back to 0. When this happened
  we need to adjust the sample position for the reset ringbuffer.
  Fixes #594136

2009-09-09 16:17:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosink.h:
  baseaudiosink: whitespace fixes

2009-09-09 16:16:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstringbuffer.c:
  ringbuffer: add more debug

2009-09-09 10:25:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/mixer.h:
  whitespace fixes

2009-09-08 17:59:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/video/gstvideosink.c:
* gst-libs/gst/video/gstvideosink.h:
  videosink: add "show-preroll-frame" property
  Add a property to disable rendering of video frames during preroll. This
  will only work for videosinks that use the new ::show_frame() vfunc instead
  of overriding basesink's preroll and render vfuncs directly.
  API: GstVideoSink:show-preroll-frame

2009-09-08 17:43:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
  ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc

2009-09-08 18:19:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/video/gstvideosink.c:
* gst-libs/gst/video/gstvideosink.h:
  video: add GstVideoSinkClass::show_frame()
  Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
  vfuncs and add some gtk-doc chunks.
  API: GstVideoSinkClass::show_frame()

2009-09-08 16:00:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/interfaces/navigation.c:
  navigation: don't do stuff inside g_return_val_if_fail() statements
  Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.

2009-08-31 20:24:22 +0200  Havard Graff <havard.graff@tandberg.com>

* gst-libs/gst/interfaces/navigation.c:
  navigation: Fix compiler warning with MSVC
  Fixes bug #594275.

2009-08-31 20:31:56 +0200  Havard Graff <havard.graff@tandberg.com>

* gst-libs/gst/rtp/gstbasertpdepayload.c:
  basertpdepayload: fix event forwarding

2009-08-31 20:36:37 +0200  Havard Graff <havard.graff@tandberg.com>

* gst-libs/gst/rtp/gstrtcpbuffer.c:
  rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
  Fixes #594258

2009-09-08 13:02:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
* gst/playback/gstplaysink.h:
  fix whitespace

2009-09-08 12:59:20 +0200  Håvard Graff <havard.graff@tandberg.com>

* gst-libs/gst/audio/gstbaseaudiosrc.c:
  baseaudiosrc: improve slave skew resync
  The old one did the mistake of not actually advancing the ringbuffer, it just
  adjusted the segbase, introducing the whole lenght of the ringbuffer as an
  extra delay in the pipeline.
  Also make sure that the resync can never go back in time, producing the same
  timestamps that has already been produced, as this can cause severe problems
  for sinks and other synching mechanisms.
  Fixes #594256

2009-09-07 17:13:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefinding: disable typefinder for headerless flac
  Disable headerless flac typefinder as long as it happily typefinds anything
  including /dev/urandom as flac and as long as it's not particularly useful
  given that such streams don't really exist in the wild.
  Also fix up some comments so that gtk-doc doesn't complain about them.

2009-09-06 15:21:43 +0300  René Stadler <mail@renestadler.de>

* sys/ximage/ximagesink.c:
  ximagesink: fix small memory leak when setting window title

2009-09-06 01:42:42 +0300  René Stadler <mail@renestadler.de>

* sys/xvimage/xvimagesink.c:
  xvimagesink: fix small memory leak when setting window title

2009-09-05 13:55:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* .gitignore:
  introspection: Add *.gir and *.typelib to .gitignore

2009-09-05 13:46:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  introduction: Fix out-of-tree build

2009-09-05 13:13:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/rtsp/Makefile.am:
  rtsp: Fix introspection build by ordering sources/headers in dependency order

2009-09-05 13:09:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/audio/Makefile.am:
  audio: Remove debug echo

2009-09-05 13:08:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/audio/Makefile.am:
  audio: Fix build of introspection data by using dependency order for the headers/sources

2009-09-05 12:31:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  introspection: Strip Gst prefix from all types/functions

2009-09-05 11:49:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/netbuffer/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  introspection: Fix build if gir-repository is not installed

2009-09-05 11:37:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/video/Makefile.am:
  video: Add gobject-introspection support

2009-09-05 11:35:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/tag/Makefile.am:
  tag: Add gobject-introspection support

2009-09-05 11:34:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/sdp/Makefile.am:
  sdp: Add gobject-introspection support

2009-09-05 11:31:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
  libs: Add nodist headers and sources to the introspection files

2009-09-05 11:28:59 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/rtsp/Makefile.am:
  rtsp: Add gobject-introspection support

2009-09-05 11:25:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/rtp/Makefile.am:
  rtp: Add gobject-introspection support

2009-09-05 11:23:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/riff/Makefile.am:
  riff: Add gobject-introspection support

2009-09-05 11:20:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/pbutils/Makefile.am:
  pbutils: Add gobject-introspection support

2009-09-05 11:17:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/netbuffer/Makefile.am:
  netbuffer: Add gobject-introspection support

2009-09-05 11:15:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/interfaces/Makefile.am:
  interfaces: Add gobject-introspection support

2009-09-05 11:04:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/fft/Makefile.am:
  fft: Add gobject-introspection support

2009-09-05 11:01:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/cdda/Makefile.am:
  cdda: Add gobject-introspection support
  This is disabled for now until gobject-introspection is fixed

2009-09-05 10:50:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/audio/Makefile.am:
  audio: Add gobject-introspection support

2009-09-05 10:40:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* configure.ac:
* gst-libs/gst/app/Makefile.am:
  app: Add gobject-introspection support

2009-09-05 10:20:24 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* common:
  Automatic update of common submodule
  From 00a859e to 19fa4f3

2009-09-04 15:48:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefind: fix midi typefinding
  We already have a audio/midi typefinder so don't override it with the midi in
  RIFF typefinder or else we fail to detect plain midi files.

2009-09-04 11:29:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: do buffering for more uris
  Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
  buffering.
  Fixes #594020

2009-09-04 07:36:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Add typefinder for Midi inside RIFF
  This is a standard Midi file format that should be supported by
  all Midi decoders and also has the mimetype audio/mid according to
  the Midi specification homepage.
  Fixes bug #594094.

2009-09-03 18:53:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppay: add some debugging

2009-09-03 17:53:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppay: handle gaps
  Add various conversion functions between time<->bytes<->rtptime that will be
  used later on.
  Refactor the min/max packet length code so that it can be used for both
  sample/frame based payloaders. Cache the returned values.
  code cleanups.
  When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
  same gap as the GStreamer timestamps gap.

2009-09-03 14:13:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppay: fix frame duration calculations
  Fix the calculation of the frame duration and rtp timestamps.
  Add some debugging

2009-09-03 14:13:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  rtppay: add some debugging

2009-09-02 19:49:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppay: use offsets for RTP timestamps
  Have a custom sample/frame function to generate an offset that the base class
  will use for generating RTP timestamps. This results in perfect RTP timestamps
  on the output buffers.
  Refactor setting metadata on output buffers.
  Add some more functionality to _flush().
  Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
  the next outgoing buffer.
  Flush the pending data on EOS.

2009-09-02 13:13:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppay: move function around

2009-09-02 13:12:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppay: fix sample duration calculation

2009-09-02 12:24:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppay: more refactoring
  Unify the sample/frame buffer handling code by making the functions plugable.

2009-09-02 12:03:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
  audiortppayload: refactor some more
  Refactor getting the packet min/max size and alignment code.
  Refactor converting bytes to time.
  change some variable to something shorter.

2009-09-02 10:46:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* win32/common/libgstrtp.def:
  audiortppayload: refactor and cleanup
  Always use the adapter when we need to fragment the incomming buffer. Use more
  modern adapter functions to avoid malloc and memcpy. The overall result is that
  the code looks cleaner while it should be equally fast and in some case avoid a
  memcpy and malloc.
  Use the adapter timestamping functions for more precise timestamps in case of
  weird disconts.
  Cache some values instead of recalculating them.
  Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
  the internal adapter.
  API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()

2009-09-03 16:56:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* common:
  Update common

2009-09-03 11:29:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  basertppay: add property to disable perfect RTP time
  Add a property to disable the generation of perfect RTP timestamps. By default
  it is active.
  API: GstBaseRTPPayload::perfect-rtptime

2009-09-02 19:47:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.c:
  basertppay: allow subclasses to influence RTP time
  Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
  which RTP timestamps are generated. Usually timestamps are created from the
  GStreamer timestamps on the buffer, which could result in imperfect RTP
  timestamps.

2009-09-02 19:44:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.h:
  basertppay: add macro to cast

2009-09-01 18:26:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiopayload: code cleanups

2009-09-01 18:08:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
  audiortppayload: don't check adapter
  the adapter is never NULL so we don't need to check it.
  Use _scale functions to avoid overflows.

2009-09-03 00:14:02 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* configure.ac:
* gst/typefind/Makefile.am:
* gst/typefind/gsttypefindfunctions.c:
  typefinding: move gio-based xdg mime typefinder from -bad to -base
  Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
  reporting a 20% probability and somesuch). Won't be registered if
  the gio plugin has been disabled via ./configure --disable-gio.

2009-09-01 15:06:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/subparse/gstsubparse.c:
  subparse: GstAdapter is not a GstObject and should be freed with g_object_unref

2009-09-01 15:02:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* sys/v4l/v4lsrc_calls.c:
  v4lsrc: fix timestamping for when we do not have a clock yet
  Should fix #559049.

2009-09-01 14:30:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* sys/v4l/v4lsrc_calls.c:
  v4lsrc: don't log not-yet-initialised integer value

2009-09-01 14:28:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* sys/v4l/v4lsrc_calls.c:
  v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
  And reflow code to be more indent friendly.

2009-09-01 10:39:52 +0200  Jonas Holmberg <jonas.holmberg@axis.com>

* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstbasertppayload.h:
  basertppayload: Make instance init faster by not reading /dev/urandom 3 times
  ... which is the default seed when creating a new GRand. Because
  GLib in older versions used buffered IO this would take a lot of time.
  Instead use the global GRand for getting random numbers and keep the
  three instance GRand for backward compatibility with a simple seed.
  Fixes bug #593284.

2009-08-31 22:48:01 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/adder/gstadder.c:
  adder: improve caps filter functionality. Fixes #590146.
  Also use the capsfilter if there is no src-peer as the caps constrain what
  we can do. Don't create any_caps as a default, as we check for NULL to skip the
  filtering. This is a (small) performance regression as we always intersect
  otherwise.

2009-08-31 11:10:55 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: Post missing plugin messages before any error messages

2009-08-28 19:06:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/cdda/gstcddabasesrc.c:
  cddabasesrc: safely handle the indexes

2009-08-28 19:06:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* win32/common/libgstrtsp.def:
  def: add new rtsp symbols

2009-08-28 14:08:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/rtp/gstbasertppayload.h:
  basertppayload: whitespace fixes.

2009-08-27 18:59:49 +0200  Marc-André Lureau <mlureau@flumotion.com>

* gst/gdp/gstgdppay.c:
  Bug 593035 - set IN_CAPS for streamheader buffer

2009-08-26 16:56:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
  playbin: The internally linked pad of the selector might be NULL in some cases

2009-08-26 16:45:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
  playbin: Fix iterate internal linked pads functions for the stream selectors
  This now used the new gst_iterator_new_single() function and as a side effect
  fixes bug #592864.

2009-08-26 09:08:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-read.c:
  riff: Add support for AVF files
  AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
  Fixes bug #593117.

2009-08-26 09:08:12 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Detect AVF files as RIFF files too
  AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
  Partially fixes bug #593117.

2009-08-21 11:51:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/elements/audioresample.c:
  audioresample: Add unit test for checking for timestamp drifts
  This also checks for perfect timestamping and offsetting.

2009-08-21 10:11:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/audioresample/gstaudioresample.c:
  audioresample: Fix drain processing
  In case we have to convert internally don't process output length input samples
  but history length input samples.

2009-08-21 10:02:05 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/elements/audioresample.c:
  audioresample: Improve debugging a bit in the unit test

2009-08-21 10:00:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/audioresample/gstaudioresample.c:
  audioresample: On the first buffer we need discont handling
  Otherwise we won't get upstream timestamps and everything and all
  output buffers would have -1 timestamps.

2009-08-21 08:23:39 +0400  Руслан Ижбулатов <lrn1986@gmail.com>

* configure.ac:
* gst/subparse/gstsubparse.c:
  subparse: Remove dependency on regex.h as it's not used anyway
  Fixes bug #592544.

2009-08-21 06:58:31 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>

* gst/audioresample/gstaudioresample.c:
  audioresample: Fix buffer overflow when pushing the drain

2009-08-21 06:57:58 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>

* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
  audioresample: Fix timestamp drift
  Fixes bug #591934.

2009-08-24 11:34:35 -0700  David Schleef <ds@schleef.org>

* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gsttextrender.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
  Remove Ronald Bultje from Authors field
  Replaced with "GStreamer maintainers
  <gstreamer-devel@lists.sourceforge.net>" or just removed,
  depending on the number of other authors.

2009-08-24 15:06:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst/playback/gstplaybin2.c:
  playbin2: fix refcounting of _get_sink()
  g_value_set_object() increases the refcount of the sink, which is not needed
  because the object should already be refcounted. Make sure this is always the
  case and use g_value_take_object().
  Fixes: #592884

2009-08-24 14:39:16 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspdefs.c:
  rtsp: Mark Transport as supporting multiple values.

2009-08-24 13:58:17 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.h:
  rtsp: Added missing Since tags.

2009-08-24 13:27:55 +0200  Eero Nurkkala <ext-eero.nurkkala at nokia.com>

* gst-libs/gst/audio/gstringbuffer.c:
  ringbuffer: Improve audiosink startup performance
  When we start the ringbuffer, immediatly continue processing samples if the
  writer prepared some for us.
  Fixes #545807

2009-08-17 11:53:43 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
  rtsp: Added new API for sending using GstRTSPWatch.
  The new API to send messages using GstRTSPWatch will first try to send the
  message immediately. Then, if that failed (or the message was not sent
  fully), it will queue the remaining message for later delivery. This avoids
  unnecessary context switches, and makes it possible to keep track of
  whether the connection is blocked (the unblocking of the connection is
  indicated by the reception of the message_sent signal).
  This also deprecates the old API (gst_rtsp_watch_queue_data() and
  gst_rtsp_watch_queue_message().)
  API: gst_rtsp_watch_write_data()
  API: gst_rtsp_watch_send_message()

2009-08-17 11:46:32 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Made gst_rtsp_watch_queue_data() thread safe.

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

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
  rtsp: Added gst_rtsp_connection_set_http_mode().
  With gst_rtsp_connection_set_http_mode() it is possible to tell the
  connection whether to allow HTTP messages to be supported. By enabling HTTP
  support the automatic HTTP tunnel support will also be disabled.
  API: gst_rtsp_connection_set_http_mode()

2009-06-16 19:35:23 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
  If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
  then just setup the base64 decoding context for the first connection.

2009-06-16 19:04:54 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Write as much as possible in gst_rtsp_source_dispatch().
  Try to write as much as possible if there are multiple messages queued.

2009-06-16 18:38:02 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
  rtsp: Add error_full callback to GstRTSPWatchFuncs.
  The error_full callback is similar to the error callback, but allows for
  better error handling. For read errors a partial message is provided to
  help an RTSP server generate a more correct error response, and for write
  errors the write queue id of the failed message is returned.

2009-08-17 18:29:17 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Made read_line() support LWS.
  Rewrote read_line() to support LWS (Line White Space), the method used by
  RTSP (and HTTP) to break long lines. Also added support for \r and \n as
  line endings (in addition to the official \r\n).

2009-08-20 14:12:50 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
  rtsp: Do not split headers which should not be split.
  From RFC 2068 section 4.2: "Multiple message-header fields with the same
  field-name may be present in a message if and only if the entire
  field-value for that header field is defined as a comma-separated list
  [i.e., #(values)]." This means that we should not split other headers which
  may contain a comma, e.g., Range and Date.

2009-08-20 14:12:09 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Parse WWW-Authenticate headers correctly.
  Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
  allows commas both to separate between multiple challenges, and within the
  challenges themself, we need to take some extra care to split these headers
  correctly.

2009-06-17 21:46:27 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Improve parse_line().
  Make parse_line() handle keys with multiple values on one line correctly.

2009-06-17 23:15:23 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Rewrote setup_tunneling().
  Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
  coded strings and duplicates of the message parsing code.

2009-08-24 10:20:16 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
  rtsp: Rewrote gen_tunnel_reply().
  Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
  than a hard coded string.

2009-08-24 10:19:35 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Ignore the Content-Length for POST requests.
  The Content-Length for POST requests with an x-sessioncookie header should
  be ignored as the length is bogus and only there to fool proxies.

2009-06-17 20:52:48 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Normalize lines (remove extra whitespace) before parsing.

2009-06-10 13:11:31 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Made parse_string() return a result.
  This will catch parsing errors when a too long string is received.

2009-06-10 11:43:31 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Improved parsing of messages.
  Do not abort message parsing as soon as there is an error. Instead parse
  as much as possible to allow a server to return as meaningful an error as
  possible.

2009-06-09 17:54:20 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c:
* gst-libs/gst/rtsp/gstrtspmessage.h:
  rtsp: Added support for HTTP messages

2009-06-09 16:22:17 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/gstrtspconnection.h:
  rtsp: Added gst_rtsp_connection_create_from_fd().
  API: gst_rtsp_connection_create_from_fd()

2009-06-09 15:27:17 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Add initial buffer support.
  The initial buffer contains data for a connection which should be used
  before starting to actually read anything from the socket.

2009-08-24 13:15:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/app/gstappsink.c:
  appsink: don't block in paused
  When we are asked to unlock we should either leave the render function or call
  the wait_preroll method to release the stream lock.
  Fixes #592657

2009-08-24 13:06:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* docs/libs/gst-plugins-base-libs-sections.txt:
  docs: fix includes for appsrc/appsink

2009-08-24 11:24:27 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspdefs.c:
* gst-libs/gst/rtsp/gstrtspdefs.h:
  rtsp: Add support for the Authentication-Info header.
  The Authentication-Info header is defined in RFC 2617 (Digest Access
  Authentication).

2009-08-20 13:11:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggmux.c:
* tests/check/pipelines/oggmux.c:
  oggmux: don't drop the streamheader field from the output caps
  Revert previous 'fix' for bug #588717 and fix it properly, whilst
  maintaining the streamheader field on the output caps. Also make
  sure we don't leak header buffers we couldn't push when downstream
  is unlinked. Add unit test for the presence of the streamheader
  field on the output caps and for the issue from bug #588717.

2009-08-18 21:45:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
  streamselector/inputselector: Use iterate internal links instead of deprecated get internal links

2009-08-19 09:31:51 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Avoid duplicated headers.
  Remove any existing Session and Date headers before adding new ones
  when sending a request. This may happen if the user of this code reuses
  a request (rtspsrc does this when resending after authorization fails).

2009-08-18 16:49:58 +0200  Peter Kjellerstedt <pkj@axis.com>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtsp: Corrected the HTTP digest authorization computation.
  Do not use sizeof() on an array passed as an argument to a function and
  expect to get anything but the size of a pointer. As a result only the
  first 4 (or 8) bytes of the response buffer were initialized to 0 in
  auth_digest_compute_response() which caused it to return a string which
  was not NUL-terminated...

2009-08-18 11:15:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: Also send SEEK events directly to a subpicture sink

2009-08-18 08:39:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gstplaysink.c:
  playsink: If a custom text sink is used, send events to it too
  Before, SEEK events would be sent to the video sink, which wouldn't
  be linked in any way to the subtitle part of the pipeline and
  subparse would never see the SEEK event. This would then seek
  the audio/video but the subtitles would continue from the old
  position instead.
  Fixes bug #591664.

2009-08-18 08:20:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Make missing plugins emit a warning message, not an error message
  The problem with an error message is, that it will stop playback completely
  while it could be that only a audio decoder plugin is missing and the video
  could be played with the available plugins.
  See bug #591677.

2009-08-13 17:42:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Post a correct error message for unknown types
  Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
  because a plugin is missing and nothing else is wrong.
  Also make it an error instead of a warning.
  Really fixes bug #591677.

2009-08-13 15:48:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/playback/gsturidecodebin.c:
  uridecodebin: Post a missing plugin message additional to the error message on unknown types
  Fixes bug #591677.

2009-08-13 10:59:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/playback/gstplaysink.c:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
  playbin2: fix error message string
  Fixes #591577.

2009-08-05 15:38:32 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst-libs/gst/riff/riff-read.c:
  riff: align API doc of gst_riff_parse_chunk with reality

2009-08-05 15:36:30 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/playback/gstdecodebin2.c:
  decodebin2: avoid assertion failure on empty/NULL caps

2009-08-12 12:09:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Also detect SVG by the <svg> starting tag
  Not all SVG images have the DOCTYPE specified.

2009-08-10 20:18:24 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtspconnection: don't use GLib-2.18 function
  g_checksum_reset() was added only in GLib 2.18, but we still require
  only 2.16, so work around that if we only have 2.16. Fixes #591357.

2009-08-10 15:40:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* tests/check/pipelines/streamheader.c:
  streamheader: Fix caps leak in the vorbisenc unit test

2009-08-10 14:14:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* tests/check/pipelines/streamheader.c:
  checks: fix stream header unit test hanging in gst_task_cleanup_all()
  Set pipelines to NULL state and unref when done.

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

* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
  rtsp: Use GLib's GChecksum instead of our own MD5 implementation

2009-08-10 03:46:39 +0300  Mart Raudsepp <leio@gentoo.org>

* gst-libs/gst/interfaces/navigation.c:
  navigation: Fix doc blurb typo for gst_navigation_send_key_event

2009-08-09 12:13:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/subparse/gstsubparse.c:
  subparse: Allow . instead of , as millisecond delimiter in srt subtitles
  Fixes bug #591207.

2009-08-08 17:51:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst-libs/gst/audio/gstaudiosrc.c:
* gst/playback/gstinputselector.c:
* gst/playback/gststreamselector.c:
  Revert inlines that cause compiler warnings and are not needed anyway

2009-08-08 15:54:57 +0200  Edward Hervey <bilboed@bilboed.com>

* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/riff/riff-media.c:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/video/gstvideofilter.c:
* gst-libs/gst/video/gstvideosink.c:
  gst-libs: Remove dead assignments and resulting unused variables.

2009-08-08 15:54:41 +0200  Edward Hervey <bilboed@bilboed.com>

* ext/alsa/gstalsadeviceprobe.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/pango/gsttextrender.c:
* ext/vorbis/vorbisenc.c:
  ext: Remove dead assignments and resulting unused variables.

2009-08-08 15:54:02 +0200  Edward Hervey <bilboed@bilboed.com>

* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstfactorylists.c:
* gst/playback/gstinputselector.c:
* gst/playback/gstplaysink.c:
* gst/playback/gststreamselector.c:
* gst/tcp/gsttcpclientsink.c:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_image.c:
* gst/videotestsrc/gstvideotestsrc.c:
  gst: Remove dead assignments and resulting unused variables

2009-08-07 13:05:42 +0200  Josep Torra <n770galaxy@gmail.com>

* docs/design/draft-va.txt:
  docs: add draft for generic introduction of video acceleration APIs idea

2009-08-07 08:53:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
  Revert "theora: Convert theoradec to libtheora 1.0 API"
  This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
  Temporarily revert until we have a workaround for debian/ubuntu
  packaging failure (see http://bugs.debian.org/528710).

2009-08-07 09:32:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Add typefinders for many game sound console formats supported by gme
  These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.

2009-07-16 11:29:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* ext/ogg/gstoggmux.c:
  oggmux: fix warning when we're not linked downstream and error out properly
  Fix caps warning when there's no element linked downstream, and pass
  not-linked flow return value correctly up the chain, so we error out
  correctly. Fixes #588717.

2009-07-31 14:59:03 -0700  David Schleef <ds@schleef.org>

* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c:
  theora: Convert theoradec to libtheora 1.0 API

2009-08-06 20:47:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextrender.c:
  textrender: Fix blitting of text over the output buffer and cairo painting

2009-08-06 09:13:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextrender.c:
  textrender: Fix endianness problems (i.e. make it work again on big endian architectures)

2009-07-31 14:27:28 +0300  Stefan Kost <ensonic@users.sf.net>

* tests/icles/test-colorkey.c:
  colorkey-test: fix xsync error

2009-07-06 23:06:50 +0300  Siarhei Siamashka <siarhei.siamashka@nokia.com>

* gst/ffmpegcolorspace/imgconvert.c:
* gst/ffmpegcolorspace/imgconvert_template.h:
  ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats

2009-07-14 12:33:29 +0300  Stefan Kost <ensonic@users.sf.net>

* gst/playback/gstplaysink.c:
  playbin2: smarter sink selection. Fixes #588523
  Don't do fallbacks if application specified a sink element. When doing the
  fallback use configured default elements instead of hardcoded linux only
  elements. Improve error messages accordingly.

2009-08-06 12:18:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>

* gst/playback/gstqueue2.c:
  queue2: post error message when pausing task if so appropriate
  If a downstream element returns an error while upstream has already
  put all data into queue2 (including EOS), upstream will no longer
  chain into queue2, so it is up to queue2 to perform some
  EOS handling / message posting in such cases.  See #589991.

2009-08-06 12:58:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* gst-libs/gst/audio/gstbaseaudiosrc.c:
  baseaudiosrc: change default slave method
  Set the default slave method to the much better skew slaving algortihm.

2009-08-06 12:01:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  textoverlay: make buffer writable
  Make the input buffer writable before changing its contents.

2009-08-06 09:55:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefinding: fix postscript typefinder probability
  Two bytes for a rare format hardly warrants MAXIMUM typefinding
  probability, POSSIBLE seems more appropriate.

2009-08-04 14:55:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  pango: Send queries from the srcpad directly to the video sinkpad

2009-08-04 14:32:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/subparse/gstsubparse.c:
  subparse: Implement POSITION query

2009-08-04 14:29:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
  subparse: Implement SEEKING query

2009-08-04 14:14:53 +0200  John Millikin <jmillikin@gmail.com>

* configure.ac:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
  tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
  Require latest core for this.
  Fixes bug #590430.

2009-08-04 12:46:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
  pango: Add support for xRGB and BGRx formats

2009-08-04 12:22:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  pango: Fix endianness issues from the pangocairo switch
  cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
  and BGRA on little endian architectures.

2009-08-04 12:11:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextoverlay.c:
  pango: Re-add shading support which was dropped by a previous patch

2009-08-04 11:58:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* configure.ac:
* ext/pango/gsttextoverlay.c:
  pango: Check if pangocairo supports vertical rendering and fix properties

2009-08-04 11:45:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gsttextrender.c:
  textrender: Use PROP_X instead of ARG_X consistently

2009-08-04 11:42:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
  pango: Some minor cleanup

2009-08-04 11:36:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* configure.ac:
  pango: Check for pangocairo instead of pangoft2

2009-08-04 11:35:10 +0200  Young-Ho Cha <ganadist@chollian.net>

* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextoverlay.h:
* ext/pango/gsttextrender.c:
* ext/pango/gsttextrender.h:
  pango: Use pango-cairo instead of pango-ft2
  pango-cairo will always use the native font rendering backend
  of the platform and provides better results.
  Fixes bug #340887.

2009-08-04 10:35:34 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Add SVG typefinder

2009-08-04 10:29:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Add postscript typefinder

2009-07-30 15:08:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Use static caps again for MPEG4 typefinding

2009-07-30 15:05:28 +0200  Arnout Vandecappelle <arnout@mind.be>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Implement better & more flexible MPEG4 typefinding
  This detects more MPEG4 streams as MPEG4.
  Fixes bug #556537.

2009-07-30 14:04:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst-libs/gst/cdda/gstcddabasesrc.c:
  cddabasesrc: Allow to specify the device name in the URI
  The allowed URI scheme is now:
  cdda://(device#)?track
  Also allow every combination of uppercase and lowercase
  characters for the protocol part.
  Fixes bug #321532.

2009-07-30 12:37:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/videoscale/gstvideoscale.c:
  videoscale: Restrict width/height to 2^15 - 1
  Otherwise integer overflows will happen, resulting in segmentation faults.
  Fixes bug #590243.

2009-07-29 14:55:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/ffmpegcolorspace/imgconvert_template.h:
  ffmpegcolorspace: Fix indention of template header

2009-07-29 14:10:35 +0200  Philip Jägenstedt <philipj@opera.com>

* gst-libs/gst/app/gstappsrc.c:
  appsrc: Clarify documentation about caps and linkage
  Fixes bug #589095.

2009-07-29 07:42:05 +0200  Benjamin Gaignard <benjamin@gaignard.net>

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Fix typefinding of SDP files
  Fixes bug #589574.

2009-07-28 20:50:06 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>

* gst/audioresample/gstaudioresample.c:
  audioresample: Take the output offsets from the input if possible
  Fixes bug #588915.

2009-07-28 15:54:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/videoscale/gstvideoscale.c:
  videoscale: Make sure to allocate enough memory for the temporary buffer
  and fix scaling of odd-height interlaced video.

2009-07-28 15:18:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/videoscale/gstvideoscale.c:
  videoscale: Fix interlaced scaling for I420
  ...and some other minor mistakes in the previous change.

2009-07-28 14:12:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* gst/ffmpegcolorspace/gstffmpegcodecmap.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/ffmpegcolorspace/imgconvert.c:
  ffmpegcolorspace: Include interlacing information in the AVPicture
  This later allows to handle interlaced AVPicture different than
  progressive ones which is needed for horizontally subsampled YUV
  formats, see bug #589242.

2009-07-28 13:55:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

* gst/videoscale/gstvideoscale.c:
* gst/videoscale/gstvideoscale.h:
  videoscale: Add support for interlaced content
  videoscale is not mixing content of two seperate fields anymore
  and does scaling on every field separately.
  Fixes bug #588761.

2009-08-06 01:44:24 +0100  Jan Schmidt <thaytan@noraisin.net>

* configure.ac:
  back to development -> 0.10.24.1

2009-08-05 02:03:44 +0100  Jan Schmidt <thaytan@noraisin.net>

* gst-plugins-base.doap:
  Add 0.10.24 release to the doap file

=== 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