path: root/ChangeLog

=== release 0.10.26 ===

2010-02-10  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* configure.ac:
	  releasing 0.10.26, "You will know when you get there"

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: (gst_alsa_mixer_find_master_mixer):
	  If there's no mixer track by the name of 'Master' or 'Front',
	  check if there's one called 'PCM' before trying the generic
	  fallback logic (fixes #506928, where we pick 'Mic' as master
	  track for the AD1984 card in a Thinkpad T61/X61 laptop).

2008-01-07 11:40:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplay-enum.*: Add enums for configuration flags.
	  Original commit message from CVS:
	  * gst/playback/gstplay-enum.c:
	  (register_gst_autoplug_select_result),
	  (gst_autoplug_select_result_get_type), (register_gst_play_flags),
	  (gst_play_flags_get_type):
	  * gst/playback/gstplay-enum.h:
	  Add enums for configuration flags.
	  * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
	  (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
	  (gst_play_bin_get_property), (no_more_pads_cb),
	  (autoplug_select_cb), (gst_play_bin_change_state):
	  Merge mode with flags.
	  Add more property getters/setters, defaults and docs.
	  Add properties to get number of audio/video/text streams.
	  Create sink object in _init so that we can always rely on it being
	  there.
	  * gst/playback/gstplaysink.c: (gst_play_sink_init),
	  (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
	  (activate_vis), (gst_play_sink_reconfigure),
	  (gst_play_sink_set_flags), (gst_play_sink_get_flags),
	  (gst_play_sink_change_state):
	  * gst/playback/gstplaysink.h:
	  Use flags to configure the sink pipelines.
	  Add tee before audio pipeline so that we can use it for visualisations.
	  Start working on integrating visualisations.
	  Remove mode, we can do everything with the flags now.
	  Add method to configue the sink pipeline.

2008-01-06 16:36:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
	  * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
	  Update to GMemoryInputStream API changes in GLib SVN and require
	  gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
	  We can also report the duration for every GSeekable, not only
	  GFileInputStream and GMemoryInputStream.

2008-01-06 14:39:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
	  Original commit message from CVS:
	  * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
	  (check_buffer_timestamp), (check_buffer_duration):
	  Turn these functions into macros so we can see right away
	  where the failure occured.

2008-01-05 22:25:05 +0000  Julien Moutte <julien@moutte.net>

	  sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
	  Original commit message from CVS:
	  2008-01-05  Julien Moutte  <julien@fluendo.com>
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
	  debugging information to understand how X calculates the stride
	  for XvImages.

2008-01-03 20:33:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
	  Original commit message from CVS:
	  * gst/volume/Makefile.am:
	  * gst/volume/gstvolume.c: (volume_choose_func),
	  (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
	  (volume_setup):
	  * gst/volume/gstvolume.h:
	  Use GstAudioFilter as base class for the volume element instead of
	  plain GstBaseTransform.

2008-01-03 07:17:05 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
	  Don't set element details for the abstract GstAudioFilter class.

2008-01-02 12:09:48 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiofilter.c:
	  (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
	  Implement get_unit_size() vmethod of GstBaseTransform.

2008-01-01 12:53:48 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
	  Original commit message from CVS:
	  * gst-libs/gst/pbutils/Makefile.am:
	  * gst-libs/gst/pbutils/pbutils.h:
	  Use glib-enum generator to have a proper enum GType for
	  GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.

2008-01-01 01:21:47 +0000  David Schleef <ds@schleef.org>

	  tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/theoraenc.c:
	  Reenable theoraenc test, which fails on the buildbot but
	  not locally.

2007-12-31 21:31:01 +0000  David Schleef <ds@schleef.org>

	  docs/: Add *-undeclared.txt to fix buildbot.
	  Original commit message from CVS:
	  * docs/libs/.cvsignore:
	  * docs/plugins/.cvsignore:
	  Add *-undeclared.txt to fix buildbot.

2007-12-31 20:45:28 +0000  David Schleef <ds@schleef.org>

	  tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Second attempt at disabling theoraenc test long enough to
	  get buildbot to compile -base.

2007-12-31 20:21:20 +0000  David Schleef <ds@schleef.org>

	  tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
	  Original commit message from CVS:
	  * tests/check/pipelines/theoraenc.c:
	  Disable theoraenc test long enough to get the buildbot to
	  compile a recent -base.

2007-12-31 13:17:29 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (stop_cb):
	  Make sure we reset the slider value to 0.0 without racing against a
	  possible g_idle that sets it to something else.

2007-12-31 00:32:53 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  sys/ximage/ximagesink.c: fix typo
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c:
	  fix typo

2007-12-30 19:21:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
	  Original commit message from CVS:
	  * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
	  * gst-libs/gst/rtsp/gstrtspdefs.h:
	  Add Location header so that we can start implementing redirects.
	  See #506025.

2007-12-29 20:55:39 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/subparse/gstssaparse.c: combine if's
	  Original commit message from CVS:
	  * gst/subparse/gstssaparse.c:
	  combine if's

2007-12-29 19:23:59 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst/subparse/gstssaparse.c: remove duplicate log message
	  Original commit message from CVS:
	  * gst/subparse/gstssaparse.c:
	  remove duplicate log message

2007-12-29 17:29:17 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/gio/gstgio.c:
	  * ext/gio/gstgio.h:
	  * ext/gio/gstgiobasesink.h:
	  * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
	  * ext/gio/gstgiobasesrc.h:
	  * ext/gio/gstgiosink.c: (gst_gio_sink_start):
	  * ext/gio/gstgiosink.h:
	  * ext/gio/gstgiosrc.h:
	  * ext/gio/gstgiostreamsink.h:
	  * ext/gio/gstgiostreamsrc.h:
	  * tests/check/pipelines/gio.c:
	  Update to latest API changes in GLib/GIO and require at least
	  gio-2.0 2.15.0 for this.
	  * ext/gio/Makefile.am:
	  Add GST_PLUGIN_LDFLAGS to LDFLAGS.

2007-12-29 16:23:23 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_chain):
	  Fix 'xyz may be used uninitialized' compiler warnings caused
	  by broken g_assert_not_reached() macro in GLib-2.15.x and don't
	  abort() in any case but properly report the error.

2007-12-28 09:00:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybin2.c: Code cleanups.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
	  (gst_play_bin_finalize), (gst_play_bin_set_uri),
	  (gst_play_bin_set_suburi), (gst_play_bin_set_property),
	  (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
	  (autoplug_select_cb), (activate_group), (deactivate_group),
	  (setup_next_source), (save_current_group),
	  (gst_play_bin_change_state):
	  Code cleanups.
	  Remove next-uri, we can use the uri property just fine.
	  Fix some crasher.
	  Unref uridecodebin when switching.
	  Fix going to READY.
	  * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
	  (gst_play_sink_init), (gst_play_sink_dispose),
	  (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
	  (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
	  (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
	  (gst_play_sink_set_property), (gst_play_sink_get_property),
	  (gen_video_chain), (gen_text_element), (gen_audio_chain),
	  (gen_vis_element), (gst_play_sink_get_mode),
	  (gst_play_sink_set_mode), (gst_play_sink_set_flags),
	  (gst_play_sink_get_flags), (gst_play_sink_request_pad),
	  (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
	  (gst_play_sink_change_state):
	  * gst/playback/gstplaysink.h:
	  Add some locking to make things threadsafe.
	  * gst/playback/test7.c: (about_to_finish_cb):
	  Fix test.

2007-12-22 12:06:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
	  Original commit message from CVS:
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
	  (gst_video_scale_get_property), (gst_video_scale_transform_caps),
	  (gst_video_scale_transform):
	  Don't claim to be able to handle/transform caps that can't really
	  be handled by the currently selected scaling method (here: RGB or
	  packed YUV with 4-tap method). Also add locking to method property.
	  * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
	  (test_basetransform_based):
	  Some test pipelines for the above (not entirely valgrind clean yet
	  apparently).

2007-12-22 05:19:00 +0000  David Schleef <ds@schleef.org>

	  gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
	  Original commit message from CVS:
	  * gst-libs/gst/video/video.c:
	  * gst-libs/gst/video/video.h:
	  Add additional RGBA and RGB-24 video formats.

2007-12-21 22:46:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
	  Original commit message from CVS:
	  * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
	  (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
	  (test_suburi_error_wrongproto), (test_missing_primary_decoder):
	  * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
	  (cddabasesrc_suite):
	  Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
	  deprecated in the future (see #498924).

2007-12-21 22:26:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gststreamselector.c: Don't leak event.
	  Original commit message from CVS:
	  * gst/playback/gststreamselector.c: (gst_selector_pad_event):
	  Don't leak event.

2007-12-20 19:43:25 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-read.c:
	  Use GST_ROUND_UP_2 macro

2007-12-20 17:13:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/.cvsignore: Ignore more.
	  Original commit message from CVS:
	  * gst/playback/.cvsignore:
	  Ignore more.

2007-12-20 10:41:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
	  * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
	  (set_active_source):
	  * gst/playback/gstplaybasebin.h:
	  * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	  (setup_sinks), (playbin_set_subtitles_visible):
	  Make switching off of subtitles work. To avoid all kind of
	  problems with unlinking of the subtitle input, we just keep
	  the subtitle inputs linked as they are and tell textoverlay
	  not to render them. Fixes #373011.
	  Other subtitle switching issues (esp. when there are both
	  external and in-stream subtitles) remain. They'll be solved
	  in playbin2.

2007-12-18 16:21:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gststreamselector.c: Init the pad segment too.
	  Original commit message from CVS:
	  * gst/playback/gststreamselector.c: (gst_selector_pad_init):
	  Init the pad segment too.

2007-12-18 15:56:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
	  (gst_audioringbuffer_open_device),
	  (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
	  (gst_audioringbuffer_release), (gst_audioringbuffer_start),
	  (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
	  (gst_audio_sink_create_ringbuffer):
	  Improve debug output.
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
	  (gst_ring_buffer_pause), (gst_ring_buffer_delay):
	  Prevent some functions from doing things and failing when the
	  ringbuffer is not yet acquired.

2007-12-18 15:32:49 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/interfaces.h:
	  Also remove interfaces.h from CVS as it is not needed anymore.

2007-12-18 15:20:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/Makefile.am:
	  interfaces.h is not used anymore so remove it from the build
	  process.

2007-12-18 01:01:23 +0000  David Schleef <ds@schleef.org>

	  gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern.  Turn on the pain.  Apologies.  It's useful for testing ve...
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  * gst/videotestsrc/gstvideotestsrc.h:
	  Add a "blink" pattern.  Turn on the pain.  Apologies.  It's useful
	  for testing vertical refresh synchronization.

2007-12-18 00:13:26 +0000  David Schleef <ds@schleef.org>

	  Add new GstVideFormat enum and write a bunch of helper functions based around it.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/video/video.c:
	  * gst-libs/gst/video/video.h:
	  Add new GstVideFormat enum and write a bunch of helper functions
	  based around it.

2007-12-17 23:41:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Makefile.am: Use new common/win32.mak.
	  Original commit message from CVS:
	  * Makefile.am:
	  Use new common/win32.mak.

2007-12-17 16:44:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_create), (gst_base_audio_src_change_state):
	  Add debug info.
	  When going from PLAYING to PAUSED, pause the ringbuffer before calling
	  the parent state change function, just like the audiosink, because the
	  parent waits for the element to finish its processing before completing
	  the state change. This makes going to PAUSED a lot snappier.
	  When going from READY to PAUSED, don't allow the ringbuffer to start
	  yet.

2007-12-17 00:01:00 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Yet another fix for broken software that produce files with an empty
	  blockalign field. Instead of completely failing, make a second attempt
	  at guessing the width/depth by looking at strf->size.

2007-12-16 23:52:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
	  (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
	  * gst-libs/gst/pbutils/install-plugins.c:
	  (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
	  * gst-libs/gst/pbutils/missing-plugins.c:
	  (gst_missing_plugin_message_get_installer_detail),
	  (gst_missing_encoder_installer_detail_new):
	  * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
	  * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
	  Turn a few g_assert_not_reached() into g_return_val_if_reached() to
	  avoid compiler warnings (#503930).

2007-12-16 23:46:16 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
	  for jpeg video streams.
	  Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
	  for the above modification.

2007-12-15 17:27:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
	  (gst_x_overlay_handle_events):
	  More guards (we don't want klass to end up being NULL).

2007-12-15 03:40:34 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/volume/gstvolume.c: (gst_volume_init):
	  Use new gst_base_transform_set_gap_aware() function as volume
	  correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
	  for this.

2007-12-14 19:06:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (msg_segment_done), (main):
	  Don't go to READY on EOS as this avoids testing of seeking and
	  restarting after EOS, use the stop button when you want to READY.
	  Don't try to do a flushing seek in segment-done, it does not make
	  sense to use this for gapless playback and is not needed.

2007-12-14 18:46:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Use separate timers for input and output rates.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
	  (reset_rate_timer), (update_in_rates), (update_out_rates),
	  (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	  (gst_queue_chain), (gst_queue_loop):
	  Use separate timers for input and output rates.
	  Pause measuring the output rate when we block for more data.
	  See #503262.

2007-12-14 16:23:06 +0000  Christian Schaller <uraeus@gnome.org>

	* gst/speexresample/Makefile.am:
	  update spec file and add two missing files for disting
	  Original commit message from CVS:
	  update spec file and add two missing files for disting

2007-12-14 09:24:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_chain):
	  Pause the timer to measure the input rate when we block because the
	  queue is filled. See #503262.

2007-12-13 15:54:00 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst-libs/gst/rtsp/gstrtspconnection.c:
	  (gst_rtsp_connection_free):
	  Close control sockets. Fixes #503440.

2007-12-13 12:31:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
	  Expose the right pad in the right place with the right element.

2007-12-13 11:40:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
	  Original commit message from CVS:
	  * gst-libs/gst/pbutils/descriptions.c: (formats):
	  Add description for 'private' dts caps (who come up with that name?).

2007-12-13 10:10:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Makefile.am: Add check-exports target and run it with 'make check'.
	  Original commit message from CVS:
	  * Makefile.am:
	  Add check-exports target and run it with 'make check'.
	  * configure.ac:
	  Be stricter about what we export in our libraries: change regexp so that
	  we only export _gst_foo(), but not __gst_foo().
	  * gst-libs/gst/cdda/base64.h: (rfc822_binary):
	  * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
	  Change internal functions to __gst_foo so they dont' get exported.
	  * win32/common/libgstaudio.def:
	  Add missing symbols.

2007-12-11 21:18:57 +0000  David Schleef <ds@schleef.org>

	* ChangeLog:
	  ChangeLog: remove conflict markers
	  Original commit message from CVS:
	  ChangeLog: remove conflict markers

2007-12-11 17:14:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
	  Original commit message from CVS:
	  * ext/gnomevfs/Makefile.am:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
	  Use gst_tag_freeform_string_to_utf8() here, which also takes
	  into account any character sets specified by the user via
	  environment variables.

2007-12-10 15:21:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audioconvert/Makefile.am: Also link to libm.
	  Original commit message from CVS:
	  * gst/audioconvert/Makefile.am:
	  Also link to libm.

2007-12-10 15:13:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  No need for floating point operations here. avoids having to link
	  against the math library too.

2007-12-10 11:16:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add one or two missing formats.  Generate ADPCM description dynamically depending on layout/format.
	  Original commit message from CVS:
	  * gst-libs/gst/pbutils/descriptions.c: (formats),
	  (format_info_get_desc):
	  * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
	  (GST_START_TEST):
	  Add one or two missing formats.  Generate ADPCM description
	  dynamically depending on layout/format.

2007-12-09 04:28:38 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
	  Original commit message from CVS:
	  * configure.ac:
	  Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.

2007-12-08 18:38:39 +0000  Robin Stocker <robin.stocker@gmx.ch>

	  gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
	  Original commit message from CVS:
	  Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
	  Some .srt files start with chunk number 0 and not chunk number 1,
	  recognise and accept those as well (fixes #502497).
	  * tests/check/elements/subparse.c: (srt_input), (srt_input0),
	  (test_src):
	  Add unit test for the above.

2007-12-06 12:08:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplay-enum.*: Add missing files.
	  Original commit message from CVS:
	  * gst/playback/gstplay-enum.c:
	  (register_gst_autoplug_select_result),
	  (gst_autoplug_select_result_get_type):
	  * gst/playback/gstplay-enum.h:
	  Add missing files.

2007-12-05 17:11:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
	  Original commit message from CVS:
	  * gst/playback/Makefile.am:
	  Group decodebin2 and uridecodebin into the same plugin so that they
	  can share the GEnumType.
	  * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
	  (_gst_select_accumulator), (gst_decode_bin_class_init),
	  (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
	  (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
	  (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
	  Add signal to sort factories instead of the more awkward autoplug-select
	  signal.
	  Modify autoplug_select so that we can try, skip or expose the
	  autopluggin of an element on a pad.
	  * gst/playback/gstfactorylists.c: (compare_ranks),
	  (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
	  (element_filter), (gst_factory_list_get_elements),
	  (gst_factory_list_debug), (gst_factory_list_filter):
	  * gst/playback/gstfactorylists.h:
	  Simplify the API, allow getting elements based on mask.
	  * gst/playback/gstplay-marshal.list:
	  Add some more marshallers.
	  * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
	  (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
	  (autoplug_select_cb), (activate_group):
	  Add support for managing non-raw sinks by providing a custom element and
	  sink list to decodebin2.
	  Try to plug non-raw sinks when decodebin2 using autoplug-select of
	  decodebin2.
	  * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
	  (gst_play_sink_set_mode), (gst_play_sink_request_pad):
	  * gst/playback/gstplaysink.h:
	  Add support for raw and non-raw sinks.
	  Add support to force sinks selected by playbin2.
	  Don't plug raw converters for non-raw sinks.
	  * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
	  (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
	  (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
	  (plugin_init):
	  Use right accumulators.
	  Proxy new signal.

2007-12-03 13:47:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	  Use runnning time as the base time instead of the timestamp.
	  Spotted by Saur on IRC.

2007-12-03 11:32:30 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Add 'WVC1' codec mapping for Windows Media VC-1 video codec.

2007-12-03 10:58:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
	  (gst_ogg_demux_read_chain):
	  If we find a new serial number but it does not contain a BOS page, make
	  sure we initialize the chain to NULL because else we will try to scan it
	  and crash. Fixes #500763

2007-11-30 17:47:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: Refactor some common code to filter factories and check caps compat.
	  Original commit message from CVS:
	  * gst/playback/Makefile.am:
	  * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
	  (get_feature_array), (decoders_filter), (sinks_filter),
	  (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
	  (gst_factory_list_filter):
	  * gst/playback/gstfactorylists.h:
	  Refactor some common code to filter factories and check caps compat.
	  * gst/playback/gstdecodebin.c:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	  (gst_decode_bin_init), (gst_decode_bin_dispose),
	  (gst_decode_bin_autoplug_continue),
	  (gst_decode_bin_autoplug_factories),
	  (gst_decode_bin_autoplug_select), (analyze_new_pad),
	  (find_compatibles):
	  * gst/playback/gstplaybin.c:
	  * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
	  (gst_play_bin_init), (gst_play_bin_finalize),
	  (autoplug_factories_cb), (activate_group):
	  * gst/playback/gstqueue2.c:
	  * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
	  (proxy_autoplug_continue_signal),
	  (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
	  (proxy_drained_signal):
	  Add some more debug info and use factor filtering code.

2007-11-26 13:19:46 +0000  Julien Moutte <julien@moutte.net>

	  configure.ac: Add QuickTime Wrapper plug-in.
	  Original commit message from CVS:
	  2007-11-26  Julien Moutte  <julien@fluendo.com>
	  * configure.ac: Add QuickTime Wrapper plug-in.
	  * gst/speexresample/gstspeexresample.c:
	  (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
	  build on Mac OS X Leopard. Incorrect printf format arguments.
	  * sys/Makefile.am:
	  * sys/qtwrapper/Makefile.am:
	  * sys/qtwrapper/audiodecoders.c:
	  (qtwrapper_audio_decoder_base_init),
	  (qtwrapper_audio_decoder_class_init),
	  (qtwrapper_audio_decoder_init),
	  (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
	  (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
	  (make_samr_magic_cookie), (open_decoder),
	  (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
	  (qtwrapper_audio_decoder_chain),
	  (qtwrapper_audio_decoder_sink_event),
	  (qtwrapper_audio_decoders_register):
	  * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
	  (fourcc_to_caps):
	  * sys/qtwrapper/codecmapping.h:
	  * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
	  (image_description_for_mp4v), (image_description_from_stsd_buffer),
	  (image_description_from_codec_data):
	  * sys/qtwrapper/imagedescription.h:
	  * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
	  (get_output_info_from_component), (dump_avcc_atom),
	  (dump_image_description), (dump_codec_decompress_params),
	  (addSInt32ToDictionary), (dump_cvpixel_buffer),
	  (DestroyAudioBufferList), (AllocateAudioBufferList):
	  * sys/qtwrapper/qtutils.h:
	  * sys/qtwrapper/qtwrapper.c: (plugin_init):
	  * sys/qtwrapper/qtwrapper.h:
	  * sys/qtwrapper/videodecoders.c:
	  (qtwrapper_video_decoder_base_init),
	  (qtwrapper_video_decoder_class_init),
	  (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
	  (fill_image_description), (new_image_description), (close_decoder),
	  (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
	  (decompressCb), (qtwrapper_video_decoder_chain),
	  (qtwrapper_video_decoder_sink_event),
	  (qtwrapper_video_decoders_register): Initial import of QuickTime
	  wrapper jointly developped by Songbird authors (Pioneers of the
	  Inevitable) and Fluendo.

2007-11-26 12:25:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: Add GAP-flag support.
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  * gst/volume/gstvolume.c:
	  * gst/volume/gstvolume.h:
	  Add GAP-flag support.

2007-11-26 08:43:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
	  Original commit message from CVS:
	  * gst/speexresample/README:
	  * gst/speexresample/arch.h:
	  * gst/speexresample/resample.c: (resampler_basic_direct_single),
	  (resampler_basic_direct_double),
	  (resampler_basic_interpolate_single),
	  (resampler_basic_interpolate_double),
	  (speex_resampler_process_native), (speex_resampler_process_float),
	  (speex_resampler_process_int),
	  (speex_resampler_process_interleaved_float),
	  (speex_resampler_process_interleaved_int),
	  (speex_resampler_get_input_latency),
	  (speex_resampler_get_output_latency):
	  * gst/speexresample/speex_resampler.h:
	  Update speex resampler to latest SVN. We're now down to only the
	  changes noted in README again.
	  * gst/speexresample/speex_resampler_wrapper.h:
	  * gst/speexresample/gstspeexresample.c:
	  (gst_speex_resample_push_drain), (gst_speex_resample_query):
	  Adjust to API changes.

2007-11-24 15:02:01 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
	  Original commit message from CVS:
	  2007-11-24  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (main): Increase the range of the
	  rate selector as I would like to test QOS behavior at higher
	  forward and reverse playback speed like say 64x.

2007-11-23 10:21:31 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
	  Original commit message from CVS:
	  * gst/speexresample/gstspeexresample.c:
	  (gst_speex_resample_update_state):
	  Only post the latency message if we have a resampler state already.

2007-11-23 10:21:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audioresample/gstaudioresample.c: Implement latency query.
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
	  (audioresample_query), (audioresample_query_type),
	  (gst_audioresample_set_property):
	  Implement latency query.

2007-11-23 10:01:33 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
	  Original commit message from CVS:
	  * gst/speexresample/gstspeexresample.c:
	  (gst_speex_resample_update_state):
	  Also post GST_MESSAGE_LATENCY if the latency changes.

2007-11-23 08:48:50 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
	  Original commit message from CVS:
	  * gst/speexresample/resample.c: (speex_resampler_get_latency),
	  (speex_resampler_drain_float), (speex_resampler_drain_int),
	  (speex_resampler_drain_interleaved_float),
	  (speex_resampler_drain_interleaved_int):
	  * gst/speexresample/speex_resampler.h:
	  * gst/speexresample/speex_resampler_wrapper.h:
	  Add functions to push the remaining samples and to get the latency
	  of the resampler. These will get added to Speex SVN in this or a
	  slightly changed form at some point too and should get merged then
	  again.
	  * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
	  (gst_speex_resample_init_state),
	  (gst_speex_resample_transform_size),
	  (gst_speex_resample_push_drain), (gst_speex_resample_event),
	  (gst_speex_fix_output_buffer), (gst_speex_resample_process),
	  (gst_speex_resample_query), (gst_speex_resample_query_type):
	  Drop the prepending zeroes and output the remaining samples on EOS.
	  Also properly implement the latency query for this. speexresample
	  should be completely ready for production use now.

2007-11-21 18:02:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_drain):
	  Our EOS time contains the base_time, _wait_eos() expects a running_time
	  so we have to subtract the base_time again before calling the function.
	  This fixes an EOS regression where the base_time was added twice and EOS
	  took longer and longer in certain situations.
	  Fixes #498767.

2007-11-21 13:04:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Expose methods for some object properties so that subclasses can more easily configure them.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
	  (gst_base_audio_sink_set_provide_clock),
	  (gst_base_audio_sink_get_provide_clock),
	  (gst_base_audio_sink_set_slave_method),
	  (gst_base_audio_sink_get_slave_method),
	  (gst_base_audio_sink_set_property),
	  (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
	  (gst_base_audio_sink_none_slaving),
	  (gst_base_audio_sink_handle_slaving):
	  * gst-libs/gst/audio/gstbaseaudiosink.h:
	  Expose methods for some object properties so that subclasses can more
	  easily configure them.
	  Added slave method none, that completely disables slaving to the
	  internal clock.
	  API: gst_base_audio_sink_set_provide_clock()
	  API: gst_base_audio_sink_get_provide_clock()
	  API: gst_base_audio_sink_set_slave_method()
	  API: gst_base_audio_sink_get_slave_method()
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_set_provide_clock),
	  (gst_base_audio_src_get_provide_clock),
	  (gst_base_audio_src_set_property),
	  (gst_base_audio_src_get_property), (gst_base_audio_src_create):
	  * gst-libs/gst/audio/gstbaseaudiosrc.h:
	  Expose methods for some object properties so that subclasses can more
	  easily configure them.
	  API: gst_base_audio_src_set_provide_clock()
	  API: gst_base_audio_src_get_provide_clock()

2007-11-21 10:18:56 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
	  Original commit message from CVS:
	  * gst/speexresample/README:
	  Add README explaining where the resampling code was taken from
	  and which changes were done.
	  * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
	  (speex_free):
	  Use g_malloc() and friends instead of malloc() to achieve higher
	  portability and define the functions inline.
	  * gst/speexresample/speex_resampler.h:
	  Add back some useless preprocessor stuff to keep the diff between
	  our version and the one from the Speex SVN repository lower.

2007-11-20 20:23:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
	  Original commit message from CVS:
	  * gst/speexresample/gstspeexresample.c:
	  (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
	  Some small cleanup and addition of a TODO item.

2007-11-20 12:56:00 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/Makefile.am: Add missing file.
	  Original commit message from CVS:
	  * gst/speexresample/Makefile.am:
	  Add missing file.

2007-11-20 07:53:56 +0000  Joe Peterson <lavajoe@gentoo.org>

	  gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
	  Original commit message from CVS:
	  Patch by: Joe Peterson <lavajoe at gentoo dot org>
	  * gst-libs/gst/sdp/gstsdpmessage.c:
	  Fix compilation on FreeBSD (Gentoo). Fixes #498228.

2007-11-20 07:47:27 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add speexresample to the docs and while at that do a make update.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-gio.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-mpeg2enc.xml:
	  * docs/plugins/inspect/plugin-musepack.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-videosignal.xml:
	  * docs/plugins/inspect/plugin-xingheader.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  * gst/speexresample/gstspeexresample.h:
	  Add speexresample to the docs and while at that do a make update.

2007-11-20 07:30:30 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
	  Original commit message from CVS:
	  * gst/speexresample/gstspeexresample.c:
	  (gst_speex_fix_output_buffer), (gst_speex_resample_process):
	  If the resampler gives less output samples than expected
	  adjust the output buffer and print a warning.

2007-11-20 07:02:45 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add resample element based on the Speex resampling algorithm.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/speexresample/arch.h:
	  * gst/speexresample/fixed_generic.h:
	  * gst/speexresample/gstspeexresample.c:
	  (gst_speex_resample_base_init), (gst_speex_resample_class_init),
	  (gst_speex_resample_init), (gst_speex_resample_start),
	  (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
	  (gst_speex_resample_transform_caps),
	  (gst_speex_resample_init_state), (gst_speex_resample_update_state),
	  (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
	  (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
	  (gst_speex_resample_event), (gst_speex_resample_check_discont),
	  (gst_speex_resample_process), (gst_speex_resample_transform),
	  (gst_speex_resample_set_property),
	  (gst_speex_resample_get_property), (plugin_init):
	  * gst/speexresample/gstspeexresample.h:
	  * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
	  (speex_free), (compute_func), (main), (sinc), (cubic_coef),
	  (resampler_basic_direct_single), (resampler_basic_direct_double),
	  (resampler_basic_interpolate_single),
	  (resampler_basic_interpolate_double), (update_filter),
	  (speex_resampler_init), (speex_resampler_init_frac),
	  (speex_resampler_destroy), (speex_resampler_process_native),
	  (speex_resampler_process_float), (speex_resampler_process_int),
	  (speex_resampler_process_interleaved_float),
	  (speex_resampler_process_interleaved_int),
	  (speex_resampler_set_rate), (speex_resampler_get_rate),
	  (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
	  (speex_resampler_set_quality), (speex_resampler_get_quality),
	  (speex_resampler_set_input_stride),
	  (speex_resampler_get_input_stride),
	  (speex_resampler_set_output_stride),
	  (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
	  (speex_resampler_reset_mem), (speex_resampler_strerror):
	  * gst/speexresample/speex_resampler.h:
	  * gst/speexresample/speex_resampler_float.c:
	  * gst/speexresample/speex_resampler_int.c:
	  * gst/speexresample/speex_resampler_wrapper.h:
	  Add resample element based on the Speex resampling algorithm.

2007-11-19 12:30:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
	  Original commit message from CVS:
	  * tests/check/libs/fft.c: (GST_START_TEST):
	  Fix scaling to really have dB instead of something else.

2007-11-19 12:08:16 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: There's a nice macro to check
	  Original commit message from CVS:
	  2007-11-19  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (main): There's a nice macro to
	  check
	  GTK version, use it.

2007-11-19 11:59:20 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: Try to support stable version of GTK.
	  Original commit message from CVS:
	  2007-11-19  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (main): Try to support stable version
	  of GTK.

2007-11-17 15:25:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/playback/: Fix the build + little README update.
	  Original commit message from CVS:
	  * gst/playback/README:
	  * gst/playback/test7.c:
	  Fix the build + little README update.

2007-11-16 16:02:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/seek.c: Add playbin2 seek pipeline.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
	  Add playbin2 seek pipeline.

2007-11-16 15:44:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: Add playbin2.
	  Original commit message from CVS:
	  * gst/playback/Makefile.am:
	  * gst/playback/gstplayback.c: (plugin_init):
	  * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
	  (eos_cb), (about_to_finish_cb), (main):
	  Add playbin2.
	  Added gapless playback example.
	  * gst/playback/gstplaybasebin.c:
	  * gst/playback/gstplaybasebin.h:
	  * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
	  * gst/playback/gstqueue2.c:
	  * gst/playback/test.c:
	  * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
	  (pad_removed_cb):
	  * gst/playback/gststreaminfo.h:
	  Change email.
	  * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
	  (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
	  (gst_play_bin_dispose), (gst_play_bin_set_uri),
	  (gst_play_bin_set_suburi), (gst_play_bin_set_property),
	  (gst_play_bin_get_property), (gst_play_bin_handle_message),
	  (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
	  (drained_cb), (unlink_group), (activate_group),
	  (setup_next_source), (gst_play_bin_change_state),
	  (gst_play_bin2_plugin_init):
	  Added raw first version of playbin2. Does chained oggs and gapless
	  playback fine. No support for raw sinks yet. No visualisations or
	  subtitles yet.
	  * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
	  (gst_play_sink_class_init), (gst_play_sink_init),
	  (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
	  (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
	  (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
	  (gst_play_sink_set_property), (gst_play_sink_get_property),
	  (post_missing_element_message), (free_chain), (add_chain),
	  (activate_chain), (gen_video_chain), (gen_text_element),
	  (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
	  (gst_play_sink_set_mode), (gst_play_sink_request_pad),
	  (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
	  (gst_play_sink_send_event), (gst_play_sink_change_state):
	  * gst/playback/gstplaysink.h:
	  Added Element that abstracts the sinks and their pipelines for playbin2.

2007-11-16 15:05:07 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
	  Original commit message from CVS:
	  * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
	  (gst_selector_pad_class_init), (gst_selector_pad_init),
	  (gst_selector_pad_finalize), (gst_selector_pad_reset),
	  (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
	  (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
	  (gst_selector_pad_chain), (gst_stream_selector_get_type),
	  (gst_stream_selector_base_init), (gst_stream_selector_class_init),
	  (gst_stream_selector_init), (gst_stream_selector_set_property),
	  (gst_stream_selector_get_linked_pad),
	  (gst_stream_selector_getcaps),
	  (gst_stream_selector_is_active_sinkpad),
	  (gst_stream_selector_activate_sinkpad),
	  (gst_stream_selector_get_linked_pads),
	  (gst_stream_selector_request_new_pad),
	  (gst_stream_selector_release_pad):
	  * gst/playback/gststreamselector.h:
	  Improve streamselector, make it select and unselect the current pad more
	  intelligently.
	  Subclass GstPad for the sinkpads of the selector.
	  Handle segments more correctly.
	  Fix caps negotiation.
	  Implement release_pad.

2007-11-16 12:51:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	  (gst_decode_group_check_if_drained), (source_pad_event_probe),
	  (remove_fakesink):
	  Add drained signal fired when decodebin finishes decoding the data.
	  Remove deprecated STATE_DIRTY message.
	  * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
	  (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
	  (analyse_source), (proxy_drained_signal), (make_decoder),
	  (source_new_pad), (value_list_append_structure_list),
	  (handle_redirect_message), (handle_message):
	  Proxy the new drained signal.
	  Handle pad removed from decodebin.
	  Handle redirect messages by sorting multiple redirections based on the
	  connection speed.

2007-11-16 11:22:09 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
	  * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
	  Fix leaking headers. Fixes #496761.

2007-11-16 11:16:58 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  sys/: Don't leak the PAR on errors. Fixes #496731.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
	  (gst_ximagesink_change_state):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
	  Don't leak the PAR on errors. Fixes #496731.

2007-11-16 10:14:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
	  (gst_tag_from_id3_user_tag):
	  Add mapping for audio cd discid tags, so we can extract
	  them from tags as well (see #347848). Also compare identifiers
	  in ID3v2 TXXX frames in a case-insensitive way to increase
	  compatibility when reading tags (discid vs. DiscID vs. DiscId).

2007-11-16 01:21:40 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-plugins-base.doap: Oops, fix the release name.
	  Original commit message from CVS:
	  * gst-plugins-base.doap:
	  Oops, fix the release name.

2007-11-16 00:44:58 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-plugins-base.doap: Add 0.10.15 release
	  Original commit message from CVS:
	  * gst-plugins-base.doap:
	  Add 0.10.15 release

2007-11-16 00:24:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS

=== release 0.10.15 ===

2007-11-16 00:14:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: releasing 0.10.15, "No need to argue"
	  Original commit message from CVS:
	  === release 0.10.15 ===
	  2007-11-15  Jan Schmidt <jan.schmidt@sun.com>
	  * configure.ac:
	  releasing 0.10.15, "No need to argue"

2007-11-16 00:04:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/en_GB.po:
	* po/es.po:
	* po/fi.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2007-11-15 21:40:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  win32/vs6/libgstfft.dsp: Convert line endings to DOS.
	  Original commit message from CVS:
	  * win32/vs6/libgstfft.dsp:
	  Convert line endings to DOS.

2007-11-15 21:14:04 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
	  Original commit message from CVS:
	  * win32/vs6/gst_plugins_base.dsw:
	  * win32/vs6/libgstfft.dsp:
	  * win32/MANIFEST:
	  Add a project file for fft plugin and remove socket
	  based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgsttag.dsp:
	  Convert line endings back to DOS.
	  Fixes #496724

2007-11-14 12:27:13 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  win32/vs6/: Convert line endings back to DOS
	  Original commit message from CVS:
	  * win32/vs6/libgstinterfaces.dsp:
	  * win32/vs6/libgstrtsp.dsp:
	  Convert line endings back to DOS

2007-11-14 11:08:48 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
	  Original commit message from CVS:
	  * gst-libs/gst/fft/kiss_fft_f32.h:
	  * gst-libs/gst/fft/kiss_fft_f64.h:
	  * gst-libs/gst/fft/kiss_fft_s16.h:
	  * gst-libs/gst/fft/kiss_fft_s32.h:
	  Don't include malloc.h which doesn't exist on Mac OSX.
	  Instead, pull in glib.h and use g_malloc/g_free for
	  consistency. Fixes: #496548

2007-11-09 15:54:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c:
	  Dont leak ghostpad. Fixes #475451.

2007-11-09 12:21:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Update some more docs and comments.
	  Original commit message from CVS:
	  * docs/design/design-decodebin.txt:
	  * gst/playback/gstdecodebin2.c: (analyze_new_pad):
	  Update some more docs and comments.

2007-11-07 16:47:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Require GIO >= 0.1.2 and adjust unit test for an API change.
	  Original commit message from CVS:
	  * configure.ac:
	  * tests/check/pipelines/gio.c: (GST_START_TEST):
	  Require GIO >= 0.1.2 and adjust unit test for an API change.

2007-11-07 15:18:54 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
	  Original commit message from CVS:
	  * ext/gio/gstgio.h:
	  Add macro to check if a stream supports seeking.
	  * ext/gio/Makefile.am:
	  * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
	  (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
	  (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
	  (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
	  (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
	  (gst_gio_base_sink_render), (gst_gio_base_sink_query),
	  (gst_gio_base_sink_set_stream):
	  * ext/gio/gstgiobasesink.h:
	  * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
	  (gst_gio_base_src_class_init), (gst_gio_base_src_init),
	  (gst_gio_base_src_finalize), (gst_gio_base_src_start),
	  (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
	  (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
	  (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
	  (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
	  * ext/gio/gstgiobasesrc.h:
	  Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
	  base classes that only require a GInputStream or GOutputStream to
	  work.
	  * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
	  (gst_gio_sink_class_init), (gst_gio_sink_init),
	  (gst_gio_sink_finalize), (gst_gio_sink_start):
	  * ext/gio/gstgiosink.h:
	  * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
	  (gst_gio_src_class_init), (gst_gio_src_init),
	  (gst_gio_src_finalize), (gst_gio_src_start):
	  * ext/gio/gstgiosrc.h:
	  Use the newly created base classes here.
	  * ext/gio/gstgio.c: (plugin_init):
	  * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
	  (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
	  (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
	  (gst_gio_stream_sink_get_property):
	  * ext/gio/gstgiostreamsink.h:
	  * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
	  (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
	  (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
	  (gst_gio_stream_src_get_property):
	  * ext/gio/gstgiostreamsrc.h:
	  Implement GstGioStreamSink and GstGioStreamSrc that have a property
	  to set the GInputStream/GOutputStream that should be used.
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/.cvsignore:
	  * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
	  (gio_testsuite), (main):
	  Add unit test for giostreamsrc and giostreamsink.

2007-11-07 11:48:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
	  Original commit message from CVS:
	  * ext/gio/gstgio.c: (plugin_init):
	  Remove nowadays unnecessary workaround for a crash.
	  * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
	  (gst_gio_sink_start), (gst_gio_sink_stop),
	  (gst_gio_sink_unlock_stop):
	  * ext/gio/gstgiosink.h:
	  * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
	  (gst_gio_src_stop), (gst_gio_src_unlock_stop):
	  * ext/gio/gstgiosrc.h:
	  Make the finalize function safer, clean up everything that could stay
	  around.
	  Reset the cancellable instead of creating a new one after cancelling
	  some operation.
	  Don't store the GFile in the element, it's only necessary for creating
	  the streams.

2007-11-06 23:35:39 +0000  Sebastien Moutte <sebastien@moutte.net>

	  gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
	  Original commit message from CVS:
	  Patch by: Sebastien Moutte  <sebastien moutte net>
	  * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
	  (gst_rtcp_unix_to_ntp):
	  * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
	  Fix some C99-isms and and a missing function that some versions of
	  MSVC don't like too much (#494346).
	  * win32/vs6/gst_plugins_base.dsw:
	  * win32/vs6/libgstaudio.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgsttag.dsp:
	  Update vs6 projects files (#494346).

2007-11-06 16:38:49 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  win32/common/: More missing symbols to export (fixes #493986).
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * win32/common/libgstaudio.def:
	  * win32/common/libgstcdda.def:
	  * win32/common/libgstinterfaces.def:
	  * win32/common/libgstnetbuffer.def:
	  * win32/common/libgstpbutils.def:
	  * win32/common/libgstrtp.def:
	  * win32/common/libgstrtsp.def:
	  * win32/common/libgsttag.def:
	  * win32/common/libgstvideo.def:
	  More missing symbols to export (fixes #493986).

2007-11-06 11:58:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/fft/gstfftf32.c:
	  * gst-libs/gst/fft/gstfftf32.h:
	  * gst-libs/gst/fft/gstfftf64.c:
	  * gst-libs/gst/fft/gstfftf64.h:
	  * gst-libs/gst/fft/gstffts16.c:
	  * gst-libs/gst/fft/gstffts16.h:
	  * gst-libs/gst/fft/gstffts32.c:
	  * gst-libs/gst/fft/gstffts32.h:
	  * tests/check/libs/fft.c: (GST_START_TEST):
	  Remove the magnitude and phase calculation functions as these have
	  very special use cases and can't even be used for the spectrum
	  element. Also adjust the docs to mention some properties of the used
	  FFT implemention, i.e. how the values are scaled. Fixes #492098.

2007-11-06 11:09:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
	  (finish_source):
	  Avoid crash when there are external subtitles (fixes #491722).

2007-11-03 10:39:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_open):
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
	  'Could not open resource for writing' is not an acceptable
	  error message when we can't open the audio device (see #492334),
	  even less so when we're trying to open it to record something.

2007-11-02 21:03:01 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  win32/common/libgstrtp.def: Add some more missing symbols (#492813).
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * win32/common/libgstrtp.def:
	  Add some more missing symbols (#492813).

2007-11-02 14:59:06 +0000  Thijs Vermeir <thijsvermeir@gmail.com>

	  tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
	  Original commit message from CVS:
	  Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
	  * tests/check/elements/audioconvert.c: (verify_convert):
	  Add check to make sure that the out caps have a channel layout
	  set on them where they should have one.

2007-11-01 13:28:59 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
	  Original commit message from CVS:
	  Patch by: Vincent Torri <vtorri at univ-evry dot fr>
	  * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
	  * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
	  Include our own _stdint.h instead of sys/types.h, makes MingW happy
	  (#492306).
	  * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
	  Use _pipe directly, GLib doesn't have a pipe() macro any longer
	  (it disappeared in GLib 2.14.0) (#492306).
	  * gst-libs/gst/sdp/Makefile.am:
	  * gst-libs/gst/sdp/gstsdpmessage.c:
	  Fix includes and LIBS for win32/Mingw (#492306).
	  * tests/examples/dynamic/addstream.c (pause_play_stream):
	  Use more portable g_usleep() instead of sleep() (#492306).

2007-11-01 12:51:57 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>

	  gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
	  Original commit message from CVS:
	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
	  * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	  (gst_ring_buffer_parse_caps):
	  Return NULL instead of an enum that happens to be 0, fixes warning
	  on MSVC (#492114).
	  * gst-libs/gst/audio/gstringbuffer.h:
	  No trailing commas in enum list (for gcc-2.9x).
	  * gst/videotestsrc/videotestsrc.c: (random_char):
	  Make information loss explicit instead of implicitly truncating to
	  eight bits via the return value.  Fixes runtime error on MSVC when
	  using the debug CRT (#492114).
	  * win32/common/config.h.in:
	  Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
	  * win32/common/libgstinterfaces.def:
	  * win32/common/libgstrtp.def:
	  Export a few more symbols (#492114).

2007-11-01 08:06:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/audio.c:
	  * gst-libs/gst/audio/audio.h:
	  Readd the deprecation guards, but preserve compilability.

2007-10-31 17:54:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
	  (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
	  Preserve channel layout when fixating the number of channels in the
	  output caps, or make sure there's a suitable channel position layout
	  set on the caps if required. Fixes #430677.

2007-10-31 17:32:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
	  Original commit message from CVS:
	  * tests/check/elements/decodebin.c: (test_text_plain_streams):
	  Make sure the pipeline really operates in push mode as it should
	  in this case.

2007-10-31 15:30:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/audio.h:
	  Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
	  compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
	  (ie. normal cvs builds) will fail.

2007-10-31 12:47:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tell gtk-doc about the deprecation guard. Apply more doc fixes.
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  * gst-libs/gst/audio/audio.c:
	  * gst-libs/gst/audio/audio.h:
	  * gst-libs/gst/interfaces/mixer.c:
	  tell gtk-doc about the deprecation guard. Apply more doc fixes.

2007-10-31 12:30:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
	  Original commit message from CVS:
	  * tests/check/libs/audio.c: (init_value_to_channel_layout),
	  (test_channel_layout_value_intersect), (audio_suite):
	  Add simple unit test to make sure GstValue intersection
	  of channel layouts works the way I think it does.

2007-10-30 20:32:14 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix the docs according to what gtk-doc complained about.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/audio/gstaudiofilter.h:
	  * gst-libs/gst/interfaces/mixer.h:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  * gst-libs/gst/rtp/gstbasertpdepayload.h:
	  * gst-libs/gst/sdp/gstsdpmessage.c:
	  Fix the docs according to what gtk-doc complained about.

2007-10-30 19:46:02 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/icles/stress-playbin.c: Fix the build.
	  Original commit message from CVS:
	  * tests/icles/stress-playbin.c:
	  Fix the build.

2007-10-30 15:54:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
	  * gst/playback/gstdecodebin2.c: (analyze_new_pad):
	  Post nice/more useful error message if we don't have a decoder for
	  the primary type.

2007-10-30 15:07:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
	  Be a bit more useful, unblock the pads after we fired the no-more-pads
	  signal so that we can use the signal to inspect and connect all pads
	  without having to keep extra state outside of decodebin.

2007-10-30 15:00:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
	  Original commit message from CVS:
	  * gst/playback/gsturidecodebin.c:
	  (gst_uri_decode_bin_autoplug_continue),
	  (gst_uri_decode_bin_class_init), (no_more_pads_full):
	  Implement default signal handler so that we return TRUE when nothing is
	  connected.

2007-10-28 11:53:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c:
	  (gst_riff_wavext_add_channel_layout),
	  (gst_riff_wave_add_default_channel_layout),
	  (gst_riff_wavext_get_default_channel_mask),
	  (gst_riff_create_audio_caps):
	  Use the ALSA channel layout as default for wav files without channel
	  layout information. This fixes playback of chan-id.wav on 5.1 systems
	  for example. Also refactor the channel layout setting a bit and add
	  more default channel orders. Fixes #489010.

2007-10-28 11:46:48 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	* ChangeLog:
	  Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
	  Original commit message from CVS:
	  (gst_riff_wavext_add_channel_layout),
	  (gst_riff_wave_add_default_channel_layout),
	  (gst_riff_wavext_get_default_channel_mask),
	  (gst_riff_create_audio_caps):
	  Use the ALSA channel layout as default for wav files without channel
	  layout information. This fixes playback of chan-id.wav on 5.1 systems
	  for example. Also refactor the channel layout setting a bit and add
	  more default channel orders. Fixes #489010.

2007-10-26 18:57:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
	  Original commit message from CVS:
	  * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
	  GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
	  -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
	  instead.

2007-10-26 12:07:14 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  update spec file
	  Original commit message from CVS:
	  update spec file

2007-10-25 17:36:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	  (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
	  (gst_decode_bin_set_subs_encoding),
	  (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
	  (gst_decode_bin_get_property), (analyze_new_pad):
	  Move subtitle encoding property to decodebin2 so that it can set the
	  property value on all elements that it autoplugs and that require it.
	  Make caps refcounting more consistent in get/set.
	  * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
	  (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
	  (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
	  (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
	  (proxy_autoplug_continue_signal),
	  (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
	  (make_decoder):
	  Proxy properties and relevant signals from the internal decodebin.
	  Make properties MT safe.

2007-10-25 15:10:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
	  Original commit message from CVS:
	  * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
	  * gst-libs/gst/tag/tags.c:
	  Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
	  GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
	  * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
	  Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
	  * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
	  (gst_tag_to_vorbis_comments):
	  Map new SORTNAME tags (these tags aren't even semi-official, so I'm
	  just mapping everything I found in the wild) (#414539).

2007-10-24 11:07:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
	  Original commit message from CVS:
	  Inspired by patch of: René Stadler <mail at renestadler dot de>
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
	  (gst_decode_bin_autoplug_continue),
	  (gst_decode_bin_autoplug_factories),
	  (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
	  (find_compatibles):
	  * gst/playback/gstplay-marshal.list:
	  Remove the autoplug-sort signal and replace it with a binding friendly
	  autoplug-select signal.
	  Add an autoplug-factories signal that can be used to generate a list of
	  factories to try to autoplug.
	  Add the GstPad to the autoplugging signal args as it might be needed to
	  make a good factory selection.
	  Fix up the marshallers for this. Fixes #407282.

2007-10-23 14:23:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gsttagdemux.c:
	  Don't abort with an assertion if we receive a seek event with
	  a start type of NONE (see launchpad bug #155878).

2007-10-22 10:21:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
	  (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
	  (gst_ximagesink_change_state), (gst_ximagesink_reset):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
	  (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
	  (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
	  Make sure that before we clean up the X resources, we shutdown and join
	  the event thread.
	  Also make sure the event thread does not shut down immediatly after
	  startup because the running variable is not yet correctly set.
	  Fixes #378770.

2007-10-16 16:48:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (new_pad), (type_found):
	  Make the window for a race in typefind and shutting down smaller until
	  we figure out the right locking here. Avoids #485753 usually.
	  * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
	  Remove unneeded lock causing a race in typefind and shutting down.
	  Fixes #485753.
	  * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
	  Also remove sinks when going to NULL because we might not complete the
	  state change to PAUSED, causing the PAUSED->READY state change not to
	  happen.

2007-10-16 15:33:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
	  Also explicitly release the ringbuffer when going to NULL because it
	  is required in the setcaps function, before the state change to PAUSED
	  completes.

2007-10-16 14:58:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/: Does what it says on the tin.
	  Original commit message from CVS:
	  * tests/icles/.cvsignore:
	  * tests/icles/Makefile.am:
	  * tests/icles/stress-playbin.c:
	  Does what it says on the tin.

2007-10-15 11:38:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
	  Fix queue negotiation. See #486758.

2007-10-12 10:52:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Actual code change to go along with:
	  Original commit message from CVS:
	  Actual code change to go along with:
	  2007-10-12  Jan Schmidt  <Jan.Schmidt@sun.com>
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
	  (gst_xvimagesink_xwindow_new),
	  (gst_xvimagesink_update_colorbalance),
	  (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
	  Fix handling of some of the X atoms. If the last parameter is True,
	  XInternAtom won't create the atom if it doesn't exist, and therefore
	  might return None. This causes X errors on Xv implementations that
	  don't provide the colour balance attributes.

2007-10-12 10:37:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Remove stray character from the changelog.
	  Original commit message from CVS:
	  Remove stray character from the changelog.

2007-10-12 10:33:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  I'm too lazy to comment this
	  Original commit message from CVS:
	  *** empty log message ***

2007-10-11 18:24:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Extract vorbis comment LICENSE tags correctly.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  * tests/check/libs/tag.c:
	  Extract vorbis comment LICENSE tags correctly.

2007-10-11 16:12:21 +0000  Jason Kivlighn <jkivlighn@gmail.com>

	  Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
	  Original commit message from CVS:
	  Patch by: Jason Kivlighn  <jkivlighn gmail com>
	  * gst-libs/gst/tag/gstid3tag.c:
	  * tests/check/libs/tag.c:
	  Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).

2007-10-10 17:01:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gsttagdemux.c:
	  Don't error out when a buggy downstream element doesn't
	  handle the newsegment event we send properly (especially
	  not without posting a meaningful error message on the
	  bus). See bug #471370 and launchpad bug #136264.

2007-10-10 15:36:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_drain):
	  Use new basesink method to make our EOS drain interruptable.

2007-10-10 09:37:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtppayloads.c:
	  Fix silly search-replace oversight.

2007-10-09 09:57:17 +0000  Laurent Glayal <spglegle@yahoo.fr>

	  gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
	  Original commit message from CVS:
	  Patch by: Laurent Glayal <spglegle at yahoo dot fr>
	  * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
	  (gst_basertppayload_set_outcaps):
	  Fix caps memleak. Fixes #484989.

2007-10-08 18:04:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
	  Fix debug output.

2007-10-08 18:02:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_create):
	  Also handle the case where there is no clock set on the audio source,
	  like in the unit tests.

2007-10-08 17:40:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtppayloads.c:
	  Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
	  to avoid compiler warnings

2007-10-08 17:12:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (type_found),
	  (gst_decode_bin_change_state):
	  * gst/playback/gstdecodebin2.c: (type_found),
	  (gst_decode_bin_change_state):
	  Don't disconnect the have_type signal because we never reconnect it
	  later on. Instead keep a variable to see if we already detected a type.

2007-10-08 10:47:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
	  (type_found):
	  Unlink the signal handler when we found the type, we're not going to do
	  anything sensible with more type_found signals anyway.

2007-10-08 06:07:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
	  Original commit message from CVS:
	  * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
	  Use GIO function to get a list of supported URI schemes instead of
	  hard coding something.

2007-10-06 16:49:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gsttagdemux.c:
	  Don't leak caps.

2007-10-06 15:04:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/Makefile.am:
	  * gst-libs/gst/tag/gsttagdemux.c:
	  * gst-libs/gst/tag/gsttagdemux.h:
	  API: add GstTagDemux base class for simple tag demuxers.
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Add GstTagDemux to docs.

2007-10-05 07:49:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtpbuffer.c:
	  (gst_rtp_buffer_get_payload_subbuffer):
	  Fix bug introduced with last commit which inverted the logic and
	  caused all buffers to be dropped. Fixes #483620.
	  Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.

2007-10-04 06:50:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtpbuffer.c:
	  Replace g_return_if_val (as it could be disabled), with regular return
	  and warning.

2007-10-03 14:51:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
	  Original commit message from CVS:
	  * tests/check/pipelines/simple-launch-lines.c:
	  Print message name and not just number.

2007-10-02 11:11:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_async_play):
	  When slaved to the clock, don't try to align a sample with the previous
	  one when going to PLAYING again.

2007-10-02 09:04:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/examples/snapshot/snapshot.c: Fix the build.
	  Original commit message from CVS:
	  * tests/examples/snapshot/snapshot.c:
	  Fix the build.

2007-10-02 07:43:57 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/gio/gstgiosink.c: Update to API changes in GIO.
	  Original commit message from CVS:
	  * ext/gio/gstgiosink.c: (gst_gio_sink_start):
	  Update to API changes in GIO.

2007-10-01 16:33:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
	  Original commit message from CVS:
	  * gst-libs/gst/sdp/gstsdpmessage.h:
	  Add RFC 3556 bandwidth modifiers.

2007-10-01 13:37:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Update documentation.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtp/gstrtppayloads.c:
	  Update documentation.

2007-10-01 13:22:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/: Added new file and header to deal with payload info.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/Makefile.am:
	  * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
	  (gst_rtp_payload_info_for_name):
	  * gst-libs/gst/rtp/gstrtppayloads.h:
	  Added new file and header to deal with payload info.
	  * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
	  (gst_rtp_buffer_default_clock_rate):
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Payload specific stuff is move to new headers.
	  Implement _default_clock rate using the new payload function.
	  * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
	  (gst_sdp_parse_line):
	  * gst-libs/gst/sdp/gstsdpmessage.h:
	  Add some more comments.

2007-10-01 10:22:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
	  (sdp_check_header), (sdp_type_find), (plugin_init):
	  Add typefind function for application/sdp.
	  Remove some old dirac typefind code that was ifdeffed out.

2007-09-29 12:04:02 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/common/libgstaudio.def: Add new exported functions.
	  Original commit message from CVS:
	  * win32/common/libgstaudio.def:
	  Add new exported functions.
	  * win32/vs6/grammar.dsp:
	  Add autogeneration and copy of some autegenerated files from win32/common
	  for rtsp library.
	  * win32/vs6/libgstaudioconvert.dsp:
	  Add gstaudioquantize.c to the build.
	  * win32/vs6/libgstinterfaces.dsp:
	  Add videoorientation.c to the build.
	  * win32/vs6/libgstriff.dsp:
	  Add libgsttag to the link libraries list.
	  * win32/vs6/libgstvolume.dsp:
	  Add liboil to the link.
	  * win32/vs6/gst_plugins_base.dsw:
	  * win32/vs6/libgstrtsp.dsp:
	  * win32/common/libgstrtsp.def:
	  Add files to build libgstrtsp library.

2007-09-29 07:01:55 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
	  Original commit message from CVS:
	  * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
	  (gst_gio_sink_set_property), (gst_gio_sink_render):
	  * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
	  (gst_gio_src_set_property):
	  Some minor cleanup and allow setting the location only when the
	  element is not playing or paused.

2007-09-26 15:14:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
	  Original commit message from CVS:
	  * tests/examples/snapshot/snapshot.c: (main):
	  Print error when pipeline failed to construct.

2007-09-25 19:06:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst-libs/gst/tag/gstid3tag.c:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  Add mappings for the new GST_TAG_COMPOSER for vorbis comments
	  and ID3v2 tags.

2007-09-25 11:54:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
	  Original commit message from CVS:
	  * gst-libs/gst/floatcast/floatcast.h:
	  Don't include config.h in an installed public header, this
	  might break compilation of applications that don't have such
	  a header and doesn't necessarily do what it's supposed to do
	  anyway (ie. check for the lrint/lrintf defines) (#442065).
	  Add docs for the various macros and document how this header
	  has to be used (link against libm, etc.); add a few FIXMEs;
	  include math.h for non-c99 code path.  Based on patch by
	  Jan Schmidt.

2007-09-25 07:50:59 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
	  Original commit message from CVS:
	  * configure.ac:
	  Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
	  of duplicating these macros in configure.ac.

2007-09-22 17:58:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/: Updated translations to 0.10.14
	  Original commit message from CVS:
	  * po/hu.po:
	  * po/sv.po:
	  * po/uk.po:
	  Updated translations to 0.10.14

2007-09-22 17:57:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/LINGUAS:
	  add languages
	  Original commit message from CVS:
	  add languages

2007-09-22 17:56:28 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/pl.po: Added Polish translation.
	  Original commit message from CVS:
	  translated by: Jakub Bogusz <qboosh@pld-linux.org>
	  * po/pl.po:
	  Added Polish translation.

2007-09-22 17:55:37 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/fi.po: Added Finnish translation.
	  Original commit message from CVS:
	  translated by: Ilkka Tuohela <hile@iki.fi>
	  * po/fi.po:
	  Added Finnish translation.

2007-09-22 17:54:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/es.po: Added Spanish translation.
	  Original commit message from CVS:
	  translated by: Jorge González González <aloriel@gmail.com>
	  * po/es.po:
	  Added Spanish translation.

2007-09-22 17:53:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/da.po: Added Danish translation.
	  Original commit message from CVS:
	  translated by: Mogens Jaeger <mogens@jaeger.tf>
	  * po/da.po:
	  Added Danish translation.

2007-09-22 17:52:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/zh_CN.po: Added Chinese (simplified) translation.
	  Original commit message from CVS:
	  translated by: Funda Wang <fundawang@linux.net.cn>
	  * po/zh_CN.po:
	  Added Chinese (simplified) translation.

2007-09-22 17:51:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/bg.po: Added Bulgarian translation.
	  Original commit message from CVS:
	  translated by: Alexander Shopov <ash@contact.bg>
	  * po/bg.po:
	  Added Bulgarian translation.

2007-09-21 18:00:24 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
	  Update hierarchy.
	  * ext/gio/gstgiosink.h:
	  * ext/gio/gstgiosrc.h:
	  Mark private fields of the instance structs private.

2007-09-21 17:31:05 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  * docs/plugins/gst-plugins-bad-plugins.args:
	  * docs/plugins/gst-plugins-bad-plugins.signals:
	  * docs/plugins/inspect/plugin-bz2.xml:
	  * docs/plugins/inspect/plugin-cdxaparse.xml:
	  * docs/plugins/inspect/plugin-dfbvideosink.xml:
	  * docs/plugins/inspect/plugin-dtsdec.xml:
	  * docs/plugins/inspect/plugin-equalizer.xml:
	  * docs/plugins/inspect/plugin-faac.xml:
	  * docs/plugins/inspect/plugin-faad.xml:
	  * docs/plugins/inspect/plugin-filter.xml:
	  * docs/plugins/inspect/plugin-freeze.xml:
	  * docs/plugins/inspect/plugin-gio.xml:
	  * docs/plugins/inspect/plugin-gsm.xml:
	  * docs/plugins/inspect/plugin-gstrtpmanager.xml:
	  * docs/plugins/inspect/plugin-h264parse.xml:
	  * docs/plugins/inspect/plugin-modplug.xml:
	  * docs/plugins/inspect/plugin-mpeg2enc.xml:
	  * docs/plugins/inspect/plugin-musepack.xml:
	  * docs/plugins/inspect/plugin-musicbrainz.xml:
	  * docs/plugins/inspect/plugin-nsfdec.xml:
	  * docs/plugins/inspect/plugin-replaygain.xml:
	  * docs/plugins/inspect/plugin-soundtouch.xml:
	  * docs/plugins/inspect/plugin-spcdec.xml:
	  * docs/plugins/inspect/plugin-spectrum.xml:
	  * docs/plugins/inspect/plugin-speed.xml:
	  * docs/plugins/inspect/plugin-tta.xml:
	  * docs/plugins/inspect/plugin-videosignal.xml:
	  * docs/plugins/inspect/plugin-xingheader.xml:
	  * docs/plugins/inspect/plugin-xvid.xml:
	  Add the GIO plugin to the docs and do a make update
	  while doing that.
	  * ext/gio/gstgiosrc.c: (gst_gio_src_start):
	  Fix a small memleak.

2007-09-21 17:07:56 +0000  René Stadler <mail@renestadler.de>

	  Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  * configure.ac:
	  * ext/Makefile.am:
	  * ext/gio/Makefile.am:
	  * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
	  (gst_gio_get_supported_protocols),
	  (gst_gio_uri_handler_get_type_sink),
	  (gst_gio_uri_handler_get_type_src),
	  (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
	  (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
	  (gst_gio_uri_handler_do_init), (plugin_init):
	  * ext/gio/gstgio.h:
	  * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
	  (gst_gio_sink_class_init), (gst_gio_sink_init),
	  (gst_gio_sink_finalize), (gst_gio_sink_set_property),
	  (gst_gio_sink_get_property), (gst_gio_sink_start),
	  (gst_gio_sink_stop), (gst_gio_sink_unlock),
	  (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
	  (gst_gio_sink_render), (gst_gio_sink_query):
	  * ext/gio/gstgiosink.h:
	  * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
	  (gst_gio_src_class_init), (gst_gio_src_init),
	  (gst_gio_src_finalize), (gst_gio_src_set_property),
	  (gst_gio_src_get_property), (gst_gio_src_start),
	  (gst_gio_src_stop), (gst_gio_src_get_size),
	  (gst_gio_src_is_seekable), (gst_gio_src_unlock),
	  (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
	  (gst_gio_src_create):
	  * ext/gio/gstgiosrc.h:
	  Add a GIO/GVFS plugin with source and sink elements. This will
	  only be enabled when --enable-experimental is given to configure
	  for now as the GIO API is not stable yet. Fixes #476916.

2007-09-21 14:37:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_push_one):
	  Fix compilation wrt printf arguments.

2007-09-20 17:38:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
	  Original commit message from CVS:
	  * examples/app/appsrc_ex.c: (main):
	  Fix compilation after changing the name of a method.

2007-09-20 14:09:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Add simple snapshot example program using appsink.
	  Original commit message from CVS:
	  * configure.ac:
	  * tests/examples/Makefile.am:
	  * tests/examples/snapshot/.cvsignore:
	  * tests/examples/snapshot/Makefile.am:
	  * tests/examples/snapshot/snapshot.c: (main):
	  Add simple snapshot example program using appsink.

2007-09-20 13:59:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
	  Original commit message from CVS:
	  * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
	  (gst_app_sink_class_init), (gst_app_sink_init),
	  (gst_app_sink_dispose), (gst_app_sink_finalize),
	  (gst_app_sink_set_property), (gst_app_sink_get_property),
	  (gst_app_sink_flush_unlocked), (gst_app_sink_start),
	  (gst_app_sink_event), (gst_app_sink_getcaps),
	  (gst_app_sink_set_caps), (gst_app_sink_get_caps),
	  (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
	  (gst_app_sink_pull_buffer):
	  * gst-libs/gst/app/gstappsink.h:
	  Add properties, signals and actions to access the element even without
	  linking to the library.
	  Fix some method names and signatures.

2007-09-20 10:37:02 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/generic/states.c: Improved state change unit test.
	  Original commit message from CVS:
	  * tests/check/generic/states.c:
	  Improved state change unit test.

2007-09-19 18:16:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Ignore registries in any format.
	  Original commit message from CVS:
	  * docs/plugins/.cvsignore:
	  * tests/check/.cvsignore:
	  Ignore registries in any format.

2007-09-19 16:09:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_chain),
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  Only copy timestamp on outgoing packets if the depayloader did not set
	  one.
	  Also copy duration on outgoing packets.

2007-09-19 15:55:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
	  (gst_basertppayload_set_outcaps):
	  Fix compilation because of missing %d in printf.
	  When fixating caps, fixate what we can and throw away all remaining
	  unfixed caps, subclasses should do something smart if they need to.

2007-09-19 12:04:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  Improve debug logs a bit and be more verbose if things go wrong.

2007-09-17 17:24:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fix a bunch of compile warnings shown with Forte.
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
	  (gst_text_overlay_set_property):
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
	  (gst_rtcp_unix_to_ntp):
	  * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
	  * gst/playback/gstqueue2.c:
	  * tests/examples/seek/seek.c: (set_scale):
	  Fix a bunch of compile warnings shown with Forte.
	  * gst/audiorate/gstaudiorate.c:
	  Always pull in config.h before including any system headers.

2007-09-17 16:22:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Also fix #476514 for queue2.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (update_buffering),
	  (gst_queue_locked_flush), (gst_queue_locked_enqueue),
	  (gst_queue_handle_sink_event), (gst_queue_chain),
	  (gst_queue_push_one), (gst_queue_sink_activate_push),
	  (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
	  Also fix #476514 for queue2.

2007-09-16 19:31:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
	  (gst_base_rtp_depayload_chain),
	  (gst_base_rtp_depayload_handle_sink_event),
	  (gst_base_rtp_depayload_push_full),
	  (gst_base_rtp_depayload_set_gst_timestamp),
	  (gst_base_rtp_depayload_change_state):
	  Remove code to deal with RTP to GST time conversion, we now just copy
	  the GST timestamp we receive to the outgoing buffers.
	  Handle segment and flushes correctly.
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	  When we have no valid input timestamp, use the previous rtp timestamp on
	  the outgoing RTP packet instead of the RTP base time.

2007-09-16 01:56:21 +0000  David Schleef <ds@schleef.org>

	  ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
	  Original commit message from CVS:
	  * ext/alsa/gstalsa.c:
	  * ext/alsa/gstalsadeviceprobe.c:
	  * ext/alsa/gstalsamixer.c:
	  * ext/alsa/gstalsasink.c:
	  * ext/alsa/gstalsasrc.c:
	  Change alsa alloca's to malloc to fix warnings on gcc-4.2.

2007-09-15 18:41:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
	  Add some debug info when negotiating caps.

2007-09-15 00:29:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
	  A buffer with an empty payload is also a valid buffer.

2007-09-14 20:52:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
	  (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
	  (gst_basertppayload_change_state):
	  Make sure we start our RTP timestamp from the random base RTP
	  timestamp even if the buffer timestamp starts from some random value.

2007-09-14 16:56:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
	  Original commit message from CVS:
	  * configure.ac:
	  * tests/examples/Makefile.am:
	  * tests/examples/dynamic/.cvsignore:
	  * tests/examples/dynamic/Makefile.am:
	  * tests/examples/dynamic/addstream.c: (create_stream),
	  (pause_play_stream), (message_received), (eos_message_received),
	  (perform_step), (main):
	  Add simple exmple app to demonstrate starting and pausing live and
	  non-live bins in a PLAYING pipeline.

2007-09-14 10:42:00 +0000  Julien Moutte <julien@moutte.net>

	  gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
	  Original commit message from CVS:
	  2007-09-14  Julien MOUTTE  <julien@moutte.net>
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
	  typefind for QCP files (RFC #3625)

2007-09-13 22:52:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_init):
	  Disable pull mode scheduling, we're not ready for it yet and it subtly
	  breaks a lot of things.

2007-09-12 17:35:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
	  Original commit message from CVS:
	  * tests/check/elements/libvisual.c:
	  Test all libvisual plugins, not just the first one; this reproduces
	  bug #450336 quite easily.  Looks like a problem with the 'jess'
	  visualisation.

2007-09-12 17:15:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/libvisual.c:
	  Add basic libvisual test case in an attempt to reproduce bug #450336.
	  Doesn't reproduce that bug, but some other crasher instead (invalid
	  free), at least with make elements/libvisual.forever and the bumscope
	  plugin on x86-64/gutsy. Leaving test disabled for now.

2007-09-12 08:38:21 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/: Printf format fixes (#476128).
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst-libs/gst/app/gstappsink.c:
	  * gst/flv/gstflvdemux.c:
	  * gst/flv/gstflvparse.c:
	  * gst/interleave/deinterleave.c:
	  * gst/switch/gstswitch.c:
	  Printf format fixes (#476128).

2007-09-11 19:07:57 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
	  * gst-libs/gst/rtsp/gstrtspconnection.c:
	  (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
	  (read_body), (gst_rtsp_connection_receive):
	  Make sure we can not cancel in the middle of receiving a message.
	  Fixes #475731.

2007-09-11 11:29:12 +0000  Josep Torra Valles <josep@fluendo.com>

	  gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
	  Original commit message from CVS:
	  Patch by: Josep Torra Valles <josep@fluendo.com>
	  * gst/playback/gstplaybasebin.c:
	  Increase upper limit for audio queue a bit; fixes preroll problem
	  with playbin and decodebin2 when playing a quicktime trailer with
	  multichannel audio via http (#464666).

2007-09-10 22:10:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_class_init), (gst_base_audio_src_init),
	  (gst_base_audio_src_provide_clock),
	  (gst_base_audio_src_set_property),
	  (gst_base_audio_src_get_property), (gst_base_audio_src_create):
	  * gst-libs/gst/audio/gstbaseaudiosrc.h:
	  Allow othe clocks than the internal clock to be used for the pipeline.
	  Add property to disable clock provide.
	  API: GstBaseAudioSrc::provide-clock

2007-09-10 12:05:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c:
	  Don't leak request pads. Fixes #475395.

2007-09-09 10:25:43 +0000  René Stadler <mail@renestadler.de>

	  sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	  (gst_ximage_buffer_class_init):
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimage_buffer_class_init):
	  Correctly chain up finalize with the parent class to prevent
	  memory leaks. Fixes #474880.

2007-09-09 04:08:48 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c: (volume_choose_func):
	  * tests/check/elements/volume.c: (GST_START_TEST):
	  Revert the latest change: floating point samples are allowed to
	  have any value, not only values in the range [-1,1]. Thanks to Andy
	  Wingo for noticing.
	  Also fix processing of int32 samples with volumes > 4 by making the
	  unity value smaller which prevents overflows.

2007-09-07 17:37:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtpbuffer.c:
	  * tests/check/libs/rtp.c:
	  Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.

2007-09-07 16:46:05 +0000  Haakon Sporsheim <haakon.sporsheim@tandberg.com>

	  gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
	  Original commit message from CVS:
	  Based on patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>
	  * gst-libs/gst/rtp/gstrtpbuffer.c:
	  Fix up GstRTPHeader helper struct so that compilers will not under
	  any circumstances add padding in between our fields, as currently
	  happens with MSVC on win32, because that would lead to us sending
	  out RTP payloads with broken RTP headers (#471194).
	  Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
	  * tests/check/Makefile.am:
	  * tests/check/libs/.cvsignore:
	  * tests/check/libs/rtp.c:
	  Add some simple unit tests for GstRTPBuffer. Some are disabled
	  because the code tested still needs fixing (set_csrc() does not work).

2007-09-07 15:05:24 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  update spec file to include latest RTSP libraries and headers and more
	  Original commit message from CVS:
	  update spec file to include latest RTSP libraries and headers and more

2007-09-07 12:41:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  win32/: Add rtsp enumtypes (#474384) and update others.
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  * win32/common/gstrtsp-enumtypes.c:
	  * win32/common/gstrtsp-enumtypes.h:
	  * win32/common/interfaces-enumtypes.c:
	  * win32/common/interfaces-enumtypes.h:
	  * win32/common/multichannel-enumtypes.c:
	  Add rtsp enumtypes (#474384) and update others.

2007-09-06 20:31:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Fix configure check for HAVE_LIBXML_HTML.
	  Original commit message from CVS:
	  * configure.ac:
	  Fix configure check for HAVE_LIBXML_HTML.

2007-09-06 12:14:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
	  Original commit message from CVS:
	  * tests/check/libs/.cvsignore:
	  Ignore more, in case the build bots work again one day.

2007-09-06 07:00:36 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  Add libgstfft, a FFT library based on Kiss FFT which is
	  Original commit message from CVS:
	  Reviewed by:  Stefan Kost  <ensonic@users.sf.net>
	  * configure.ac:
	  * gst-libs/gst/Makefile.am:
	  * gst-libs/gst/fft/Makefile.am:
	  * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
	  * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
	  * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
	  * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
	  * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
	  * gst-libs/gst/fft/gstfft.h:
	  * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
	  (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
	  (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
	  * gst-libs/gst/fft/gstfftf32.h:
	  * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
	  (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
	  (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
	  * gst-libs/gst/fft/gstfftf64.h:
	  * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
	  (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
	  (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
	  * gst-libs/gst/fft/gstffts16.h:
	  * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
	  (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
	  (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
	  * gst-libs/gst/fft/gstffts32.h:
	  * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
	  (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	  (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
	  (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
	  * gst-libs/gst/fft/kiss_fft_f32.h:
	  * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
	  (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	  (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
	  (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
	  * gst-libs/gst/fft/kiss_fft_f64.h:
	  * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
	  (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	  (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
	  (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
	  * gst-libs/gst/fft/kiss_fft_s16.h:
	  * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
	  (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
	  (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
	  (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
	  * gst-libs/gst/fft/kiss_fft_s32.h:
	  * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
	  (kiss_fftr_f32), (kiss_fftri_f32):
	  * gst-libs/gst/fft/kiss_fftr_f32.h:
	  * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
	  (kiss_fftr_f64), (kiss_fftri_f64):
	  * gst-libs/gst/fft/kiss_fftr_f64.h:
	  * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
	  (kiss_fftr_s16), (kiss_fftri_s16):
	  * gst-libs/gst/fft/kiss_fftr_s16.h:
	  * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
	  (kiss_fftr_s32), (kiss_fftri_s32):
	  * gst-libs/gst/fft/kiss_fftr_s32.h:
	  * gst-libs/gst/fft/kiss_version:
	  * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	  * pkgconfig/gstreamer-plugins-base.pc.in:
	  Add libgstfft, a FFT library based on Kiss FFT which is
	  BSD licensed. Supported sample formats are int16, int32,
	  float and double. For those formats a real FFT and IFFT
	  can be done, different windowing functions can be applied
	  and functions for extracting the magnitude and phase exist.
	  Fixes #468619.
	  * docs/libs/Makefile.am:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Integrate libgstfft into the docs.
	  * tests/check/Makefile.am:
	  * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
	  Add unit tests for libgstfft, currently only testing the FFT.
	  Unit tests for IFFT will follow soon.

2007-09-05 23:07:40 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
	  (gst_sdp_message_init), (gst_sdp_message_uninit),
	  (is_multicast_address), (gst_sdp_message_as_text),
	  (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
	  (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
	  (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
	  (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
	  (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
	  (gst_sdp_media_init), (gst_sdp_media_uninit),
	  (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
	  (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
	  (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
	  (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
	  (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
	  * gst-libs/gst/sdp/gstsdpmessage.h:
	  Separate INIT_ARRAY() and related macros into two versions, one for
	  structures and one for pointers (e.g., INIT_ARRAY() and
	  INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
	  lists of emails and phone numbers.
	  Add missing const as appropriate.
	  Change all gint to guint since they all actually represent unsigned
	  values.
	  Do not use time as a variable name as it shadows the global time().
	  Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
	  Actually implement gst_sdp_message_add_time().
	  Make gst_sdp_message_add_time() take repeat times as an argument.
	  Store repeat times in GstSDPTime as a GArray rather than as gchar**.
	  Corrected the definition of gst_sdp_media_get_bandwidth() (was
	  misspelled as badwidth).
	  gst-indented and a little clean up. Fixes #471067.

2007-09-05 21:20:12 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c: (volume_choose_func),
	  (volume_process_double), (volume_process_double_clamp),
	  (volume_process_float_clamp):
	  Correctly clamp float/double samples in the [-1.0,1.0] range to
	  prevent weird effects.
	  * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	  Add unit tests for all samples types that had none before.

2007-09-05 14:09:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtpbuffer.c:
	  Need to include stdlib.h for abs() here too.

2007-09-05 14:01:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gststreaminfo.c: Fix build.
	  Original commit message from CVS:
	  * gst/playback/gststreaminfo.c:
	  Fix build.

2007-09-05 10:32:09 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
	  Original commit message from CVS:
	  * gst/playback/gststreaminfo.c:
	  Clean up some half-disabled code and comment.

2007-09-04 16:18:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_payload_audio_handle_event):
	  Return FALSE from the event handler to let the parent class handle the
	  event.
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
	  Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  Bump the MTU to 1400.

2007-09-04 01:50:55 +0000  Johan Dahlin <johan@gnome.org>

	  gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
	  Original commit message from CVS:
	  2007-09-03  Johan Dahlin  <jdahlin@async.com.br>
	  * gst/typefind/gsttypefindfunctions.c (plugin_init):
	  Add an audio/x-nsf typefind function for the nsfdec element.

2007-09-03 20:46:38 +0000  Renato Filho <renato.filho@indt.org.br>

	  gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c:
	  Included "myth://" on stream_uris list for enable buffering to mythtv files

2007-09-03 19:31:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Fix parsing of RB blocks.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
	  (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
	  (gst_rtcp_unix_to_ntp):
	  * gst-libs/gst/rtp/gstrtcpbuffer.h:
	  Fix parsing of RB blocks.
	  Fix docs.
	  Added helper functions to convert to/from UNIX and NTP time.
	  API: gst_rtcp_ntp_to_unix()
	  API: gst_rtcp_unix_to_ntp()
	  * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
	  (gst_rtp_buffer_get_header_len),
	  (gst_rtp_buffer_get_extension_data),
	  (gst_rtp_buffer_get_payload_subbuffer),
	  (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
	  (gst_rtp_buffer_ext_timestamp):
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Fix some more docs.
	  Implement handling of packets with extensions.
	  Fix padding check in _validate().
	  Added function to get extension data.
	  API: gst_rtp_buffer_get_header_len()
	  API: gst_rtp_buffer_get_extension_data()

2007-09-03 19:19:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_class_init),
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  Add some more docs for the queue-delay property and fix a typo in a
	  comment.
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	  Fix typo.

2007-09-03 19:17:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
	  (gst_base_audio_sink_change_state):
	  When skew slaving, try to hover around the middle of a segment so that
	  we at most drift by half a segment.
	  If we are aligning in the oposite direction of the clock skew, we don't
	  have to resync.

2007-08-31 21:07:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_setcaps),
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  Be less silly with the segment start, just apply the clock-base to the
	  timestamp.

2007-08-31 15:58:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_class_init),
	  (gst_base_rtp_depayload_finalize),
	  (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	  (gst_base_rtp_depayload_handle_sink_event),
	  (gst_base_rtp_depayload_set_gst_timestamp),
	  (gst_base_rtp_depayload_change_state):
	  * gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Deprecate the queue handling thread thing and remove the code.
	  Use new method to calculate the extended timestamp.

2007-08-31 15:21:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtcpbuffer.c:
	  (gst_rtcp_packet_sdes_copy_entry):
	  Use g_strndup which does exactly what we want.
	  * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
	  (gst_rtp_buffer_ext_timestamp):
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Add helper function to compare seqnums.
	  Add helper function to calculate extended timestamps.
	  API: gst_rtp_buffer_compare_seqnum()
	  API: gst_rtp_buffer_ext_timestamp()

2007-08-30 21:59:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtcpbuffer.c:
	  (gst_rtcp_packet_sdes_get_entry),
	  (gst_rtcp_packet_sdes_copy_entry):
	  * gst-libs/gst/rtp/gstrtcpbuffer.h:
	  Fix and document SDES item data function.
	  Add new function that makes a proper copy of SDES item data.
	  API: gst_rtcp_packet_sdes_copy_entry()

2007-08-30 07:29:55 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/Makefile.am:
	  The tcp and subparse plugins are under gst, but not totaly free of
	  dependencies. Handle selection inconfigure.ac, so that they show up
	  on the final list of what is build and what is not. Maybe they should
	  better be moved to ext.

2007-08-30 06:58:46 +0000  Daniel Díaz <yosoy@danieldiaz.org>

	  Check if libxml provides HTML parser which subparse needs.
	  Original commit message from CVS:
	  Patch by: Daniel Díaz  <yosoy@danieldiaz.org>
	  * configure.ac:
	  * gst/Makefile.am:
	  Check if libxml provides HTML parser which subparse needs.
	  Fixes #451970.

2007-08-29 14:22:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
	  Original commit message from CVS:
	  * ext/alsa/gstalsa.c:
	  Fix typo and compilation on big endian systems.

2007-08-29 12:16:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
	  Original commit message from CVS:
	  * gst/subparse/gstssaparse.c:
	  Convert SSA newline codes into actual newline characters (#470766).

2007-08-28 14:58:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  API: also add gst_install_plugins_supported() while we're at it (see #470456).
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/pbutils/install-plugins.c:
	  * gst-libs/gst/pbutils/install-plugins.h:
	  * tests/check/libs/pbutils.c:
	  API: also add gst_install_plugins_supported() while we're at it
	  (see #470456).

2007-08-28 14:23:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/pbutils/missing-plugins.c:
	  * gst-libs/gst/pbutils/missing-plugins.h:
	  * tests/check/libs/pbutils.c:
	  API: add gst_missing_*_installer_detail_new() convenience API so
	  that applications that know exactly what they're missing can request
	  installer detail strings for those items directly instead of having
	  to first create a dummy missing-plugin message and then get the
	  installer detail string from that.  Fixes #470456.

2007-08-27 11:59:56 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (close_pad_link):
	  We need to set up delayed-linking whenever the caps are non-fixed,
	  not just when there are multiple types - use gst_pad_is_fixed()
	  to test.

2007-08-26 14:14:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
	  Original commit message from CVS:
	  * gst-libs/gst/pbutils/missing-plugins.c:
	  (gst_missing_plugin_message_get_installer_detail):
	  Add missing separator in PID fallback case.

2007-08-24 15:28:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
	  Original commit message from CVS:
	  * ext/alsa/Makefile.am:
	  There is no GST_PLUGINS_BASE_LIBS defined.
	  * ext/alsa/gstalsa.c:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
	  Add support for ALSA 24-bit formats.
	  snd_pcm_delay can return an error code, especially
	  during XRUNS. In that case, the best we can do is assume
	  delay = 0.
	  * gst/audioconvert/Makefile.am:
	  Add flags from -base before any more-remote dependencies.

2007-08-23 20:45:45 +0000  Davyd <davyd@madeley.id.au>

	  gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
	  Original commit message from CVS:
	  Based on a patch by: Davyd <davyd at madeley dot id dot au>
	  * gst/volume/gstvolume.c: (volume_choose_func),
	  (volume_update_real_volume), (gst_volume_set_volume),
	  (gst_volume_init), (volume_process_int32),
	  (volume_process_int32_clamp), (volume_process_int24),
	  (volume_process_int24_clamp), (volume_process_int16),
	  (volume_process_int16_clamp), (volume_process_int8),
	  (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
	  * gst/volume/gstvolume.h:
	  Add support for int32, int24 and int8 to the volume element.
	  Fixes #445529.

2007-08-23 12:37:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/examples/Makefile.am: Fix even more.
	  Original commit message from CVS:
	  * tests/examples/Makefile.am:
	  Fix even more.

2007-08-23 10:58:42 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/libs/Makefile.am:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  * ext/gnomevfs/gstgnomevfssrc.h:
	  * gst-libs/gst/Makefile.am:
	  * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	  * pkgconfig/gstreamer-plugins-base.pc.in:
	  * sys/v4l/v4lsrc_calls.c:
	  * tests/examples/Makefile.am:
	  * win32/common/config.h:
	  Revert unwanted commit. many thanks to moap. I want a fix for
	  https://thomas.apestaart.org/moap/trac/ticket/239

2007-08-23 08:33:43 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	* ChangeLog:
	* configure.ac:
	* docs/libs/Makefile.am:
	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	* ext/gnomevfs/gstgnomevfssrc.c:
	* ext/gnomevfs/gstgnomevfssrc.h:
	* gst-libs/gst/Makefile.am:
	* gst-libs/gst/audio/gstaudiofilter.h:
	* gst/typefind/gsttypefindfunctions.c:
	* gst/volume/gstvolume.c:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	* sys/v4l/v4lsrc_calls.c:
	* tests/examples/Makefile.am:
	* win32/common/config.h:
	  Original commit message from CVS: reviewed by: <delete if not using a buddy> patch by: <delete if not someone else's patch> * configure.ac: * docs/libs/Makefile.am: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * ext/gnomevfs/gstgnomevfssrc.c: * ext/gnomevfs/gstgnomevfssrc.h: * gst-libs/gst/Makefile.am: * gst-libs/gst/audio/gstaudiofilter.h: * gst/typefind/gsttypefindfunctions.c: * gst/volume/gstvolume.c: * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: * pkgconfig/gstreamer-plugins-base.pc.in: * sys/v4l/v4lsrc_calls.c: * tests/examples/Makefile.am: * win32/common/config.h:

2007-08-22 15:29:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/audio.c: Clarify the docs a little.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/audio.c:
	  Clarify the docs a little.

2007-08-22 11:20:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c:
	  Enable liboil for float and add more details about problems with
	  int16.

2007-08-21 15:43:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
	  Original commit message from CVS:
	  * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
	  Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.

2007-08-21 12:08:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
	  When calculating the first timestamp of the buffers, don't go below 0
	  and clip the samples because the offset was on the eos page.
	  Fixes #466717.

2007-08-21 11:42:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
	  (gst_ogg_demux_collect_chain_info):
	  Also submit the eos page when trying to find the first timestamp.
	  See #466717.

2007-08-17 15:24:43 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/audio.h:
	  Use gst_util_uint64_scale() instead of doing the math
	  with double for GST_FRAMES_TO_CLOCK_TIME() and
	  GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
	  prevents rounding errors. Fixes #467667.

2007-08-17 13:42:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
	  Original commit message from CVS:
	  * gst-libs/gst/rtsp/gstrtspconnection.c:
	  (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
	  (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
	  * gst-libs/gst/rtsp/gstrtspconnection.h:
	  Small cleanups.
	  On shutdown, don't read the control socket yet.
	  Set timeout value correctly in all cases.
	  Add function to check if the server accepts reads or writes.
	  API: gst_rtsp_connection_poll()
	  * gst-libs/gst/rtsp/gstrtspdefs.h:
	  Fix compilation with -pedantic.
	  Add enum for _poll.

2007-08-16 17:11:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
	  Original commit message from CVS:
	  * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
	  Override the preroll vmethod instead of overriding the render method
	  twice.

2007-08-16 16:06:21 +0000  Olivier Crete <tester@tester.ca>

	  gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
	  Original commit message from CVS:
	  Patch by: Olivier Crete  <tester at tester ca>
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
	  (gst_basertppayload_getcaps):
	  * gst-libs/gst/rtp/gstbasertppayload.h:
	  Add getcaps vfunc to basertppayload. See #465146.

2007-08-16 11:20:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
	  Only post buffering messages when we are a stream.

2007-08-15 17:05:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/pbutils/: Small docs fix and addition.
	  Original commit message from CVS:
	  * gst-libs/gst/pbutils/install-plugins.c:
	  * gst-libs/gst/pbutils/missing-plugins.c:
	  Small docs fix and addition.

2007-08-14 17:47:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/app/gstappsink.c: Don't use new API.
	  Original commit message from CVS:
	  * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
	  Don't use new API.

2007-08-14 17:38:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/app/gstappsink.*: Make love to appsink.
	  Original commit message from CVS:
	  * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
	  (gst_app_sink_class_init), (gst_app_sink_dispose),
	  (gst_app_sink_flush_unlocked), (gst_app_sink_start),
	  (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
	  (gst_app_sink_render), (gst_app_sink_get_caps),
	  (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
	  (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
	  * gst-libs/gst/app/gstappsink.h:
	  Make love to appsink.
	  Make it support pulling of the preroll buffer.
	  Add docs and debug statements.
	  Fix some races wrt to EOS handling and stopping.
	  Implement getcaps.
	  Implement FLUSHING.
	  API: gst_app_sink_pull_preroll()

2007-08-13 15:37:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/icles/: Add a dumb little test for textoverlay alignments.
	  Original commit message from CVS:
	  * tests/icles/.cvsignore:
	  * tests/icles/Makefile.am:
	  * tests/icles/test-textoverlay.c:
	  Add a dumb little test for textoverlay alignments.

2007-08-13 15:26:54 +0000  Dan Williams <dcbw@redhat.com>

	  ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
	  Original commit message from CVS:
	  Patch by: Dan Williams  <dcbw redhat com>
	  * ext/pango/gsttextoverlay.c:
	  * ext/pango/gsttextoverlay.h:
	  API: add "line-alignment" property (#459334). Add gtk-doc blurb for
	  "silent" property so there's a Since tag in the API reference.

2007-08-13 11:21:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	  fix ... by: lines
	  Original commit message from CVS:
	  fix ... by: lines

2007-08-12 16:30:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  (gst_basertppayload_set_outcaps):
	  * gst-libs/gst/rtp/gstbasertppayload.h:
	  Improve caps negotiation so that downstream elements can confiure
	  certain RTP properties by fixing them on the caps. See #465146.
	  Add docs.

2007-08-11 12:39:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  * gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Mark as deprecated some macros which were presumably meant to be
	  private API and accidentally exposed in the public header file.
	  Also actually _init() lock (only works at the moment because the
	  struct is zeroed out when created and the initial values in the
	  mutex struct are zeroes too). (#459585)

2007-08-10 17:35:52 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/Makefile.am: Remove cruft and do some cleanups.
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  Remove cruft and do some cleanups.
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  Prepare for comming gtkdoc features (rebase against online docs).

2007-08-10 13:55:44 +0000  Michael Smith <msmith@xiph.org>

	  gst/audiorate/gstaudiorate.c: Debug output fixes.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  Debug output fixes.
	  * tests/check/elements/audiorate.c: (do_perfect_stream_test),
	  (GST_START_TEST):
	  Change the number of buffers used; 500 is too many and leads to
	  timeouts.

2007-08-10 10:08:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Printf format fixes (#465028).
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c:
	  * gst/videorate/gstvideorate.c:
	  Printf format fixes (#465028).

2007-08-09 15:44:02 +0000  Michael Smith <msmith@xiph.org>

	  gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  If we have a large (> 1 second) discontinuity, push a series of
	  smaller buffers rather than a single very large buffer. Avoids
	  unreasonably large single buffer allocations when encountering a
	  large gap.
	  * tests/check/elements/audiorate.c: (GST_START_TEST),
	  (audiorate_suite):
	  Add a test for this.

2007-08-09 12:06:43 +0000  Josep Torra Valles <josep@fluendo.com>

	  gst/playback/gstplaybasebin.c: Fixes: #465015
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (group_commit),
	  (queue_remove_probe), (queue_threshold_reached):
	  Patch by: Josep Torra Valles <josep@fluendo.com>
	  Fixes: #465015
	  Make sure we remove the check_queues buffer probe from the
	  correct queue to avoid racily going back to "buffering 99%" when
	  buffering is actually complete.
	  Also, fix the spelling of Josep's surname in the ChangeLog.

2007-08-09 11:37:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/ogg/gstoggmux.c: Do not leak oggmux instance.
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c:
	  Do not leak oggmux instance.
	  * ext/vorbis/vorbisenc.c:
	  Also log values.

2007-08-09 10:51:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/: Updated translations.
	  Original commit message from CVS:
	  * po/hu.po:
	  * po/it.po:
	  * po/nl.po:
	  * po/uk.po:
	  * po/vi.po:
	  Updated translations.

2007-08-08 16:07:21 +0000  Yang Hong <hongyang@redflag-linux.com>

	  ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
	  Original commit message from CVS:
	  patch by: Yang Hong <hongyang@redflag-linux.com>
	  * ext/pango/gsttextoverlay.c:
	  * ext/pango/gsttextoverlay.h:
	  Add 'silent' property to GstTimeOverlay. Fixes #462979

2007-08-08 15:05:22 +0000  Josep Torre Valles <josep@fluendo.com>

	  Add connection-speed property. Fixes #464690.
	  Original commit message from CVS:
	  Patch by: Josep Torre Valles <josep@fluendo.com>
	  * docs/plugins/gst-plugins-base-plugins.args:
	  * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
	  (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
	  (gst_uri_decode_bin_get_property), (gen_source_element):
	  Add connection-speed property. Fixes #464690.

2007-08-07 15:13:46 +0000  Damien Lespiau <damien.lespiau@gmail.com>

	  Fix compilation on windows. Fixes #464320.
	  Original commit message from CVS:
	  Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
	  * configure.ac:
	  * gst-libs/gst/rtsp/Makefile.am:
	  * gst-libs/gst/rtsp/gstrtspconnection.c:
	  (gst_rtsp_connection_connect):
	  Fix compilation on windows. Fixes #464320.

2007-08-07 14:14:54 +0000  Josep Torre Valles <josep@fluendo.com>

	  gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
	  Original commit message from CVS:
	  Patch by: Josep Torre Valles <josep@fluendo.com>
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	  (gst_play_base_bin_init), (queue_threshold_reached),
	  (gen_source_element), (setup_substreams),
	  (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
	  (gst_play_base_bin_get_streaminfo_value_array):
	  * gst/playback/gstplaybasebin.h:
	  * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	  (gst_play_bin_set_property), (gst_play_bin_get_property),
	  (gst_play_bin_handle_redirect_message):
	  Move connection-speed property from playbin to playbasebin so that we
	  can also configure it in source elements that have the connection-speed
	  property. Fixes #464028.
	  Add some debug info here and there.

2007-08-06 16:42:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
	  Properly respond to conversion queries. Fixes #464079.

2007-08-03 19:53:11 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
	  (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
	  (gst_audio_test_src_init_sine_table),
	  (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  Add float/double and int32 support to audiotestsrc. Fixes #460422.
	  Also set the default volume to the default value specified in the
	  GParamSpec.

2007-08-03 19:40:14 +0000  Jens Granseuer <jensgr@gmx.net>

	  gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
	  Original commit message from CVS:
	  Patch by: Jens Granseuer <jensgr at gmx dot net>
	  * gst/audioconvert/gstaudioquantize.c:
	  Fix C89 incompatibilities and spelling of explanations. Fixes #463215.

2007-08-03 15:44:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
	  Original commit message from CVS:
	  * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
	  Add rdt manager for rdt transport.
	  Fix parsing of RDT transport.

2007-08-03 14:43:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS

=== release 0.10.14 ===

2007-08-03 14:41:46 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* gst-plugins-base.doap:
	* win32/common/config.h:
	  Release 0.10.14
	  Original commit message from CVS:
	  Release 0.10.14

2007-08-03 14:24:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/de.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2007-07-27 17:37:19 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
	  Original commit message from CVS:
	  * tests/check/libs/audio.c: (GST_START_TEST):
	  Fix the test to reflect the behaviour of gst_audio_clip_buffer.

2007-07-27 17:10:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/audio.c:
	  When clipping a buffer with no timestamp, assume it is
	  within the segment without warnings.
	  Fixes: #460978

2007-07-27 11:16:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
	  Original commit message from CVS:
	  * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
	  Fire the signal on the object, not the interface.

2007-07-27 09:17:19 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
	  Original commit message from CVS:
	  * gst-libs/gst/rtsp/.cvsignore:
	  Ber. Don't include the full path, idiot.

2007-07-27 08:29:29 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
	  Original commit message from CVS:
	  * gst-libs/gst/rtsp/.cvsignore:
	  Ignore generated files.

2007-07-26 19:57:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/Makefile.am:
	  * gst-libs/gst/interfaces/interfaces-marshal.list:
	  * gst-libs/gst/interfaces/rtspextension.c:
	  * gst-libs/gst/interfaces/rtspextension.h:
	  * gst-libs/gst/rtsp/Makefile.am:
	  * gst-libs/gst/rtsp/gstrtsp.h:
	  * gst-libs/gst/rtsp/gstrtspextension.c:
	  (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
	  (gst_rtsp_extension_detect_server),
	  (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
	  (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
	  (gst_rtsp_extension_configure_stream),
	  (gst_rtsp_extension_get_transports),
	  (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	  * gst-libs/gst/rtsp/gstrtspextension.h:
	  * gst-libs/gst/rtsp/rtsp-marshal.list:
	  Move the rtspextension.h interface into gstrtspextension.h
	  as part of libgstrtsp instead of libgstinterfaces, because it's
	  only for use within plugins, not applications.
	  Add stuff to do the enum & marshal generation needed in libgstrtsp now.
	  Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
	  signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
	  is abstract.

2007-07-26 15:48:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/interfaces/: Fix marshaller for the send signal.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/Makefile.am:
	  * gst-libs/gst/interfaces/interfaces-marshal.list:
	  * gst-libs/gst/interfaces/rtspextension.c:
	  (gst_rtsp_extension_iface_init),
	  (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	  * gst-libs/gst/interfaces/rtspextension.h:
	  Fix marshaller for the send signal.
	  Add URL to stream selection interface method.

2007-07-26 15:35:43 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/Makefile.am:
	  Pull in our dependencies from -base before those from outside.

2007-07-26 14:33:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  API: gst_rtsp_base64_decode_ip()
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
	  * gst-libs/gst/rtsp/gstrtspbase64.h:
	  API: gst_rtsp_base64_decode_ip()
	  Added function to decode Base64 in-place.

2007-07-26 14:08:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/libs/.cvsignore: Ignore the mixer test binary.
	  Original commit message from CVS:
	  * tests/check/libs/.cvsignore:
	  Ignore the mixer test binary.

2007-07-26 10:00:37 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
	  Gratuitous comment change to trigger a rebuild on the buildbots.

2007-07-25 18:20:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
	  Original commit message from CVS:
	  * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
	  (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
	  (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
	  (gst_sdp_media_get_format), (gst_sdp_media_get_information),
	  (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
	  (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
	  (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
	  (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
	  (gst_sdp_media_get_attribute_val):
	  * gst-libs/gst/sdp/gstsdpmessage.h:
	  Constify args where we can.

2007-07-25 18:18:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/Makefile.am:
	  * gst-libs/gst/interfaces/rtspextension.c:
	  (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
	  (gst_rtsp_extension_detect_server),
	  (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
	  (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
	  (gst_rtsp_extension_configure_stream),
	  (gst_rtsp_extension_get_transports),
	  (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
	  * gst-libs/gst/interfaces/rtspextension.h:
	  Move interface for RTSP extensions from -good to here.
	  Added helper methods to invoke interface methods.

2007-07-25 11:22:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Fix some more RTSP docs.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtsp/gstrtspdefs.h:
	  * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
	  (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
	  (gst_rtsp_message_init_response),
	  (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
	  (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
	  (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
	  (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
	  (gst_rtsp_message_get_body), (dump_key_value):
	  * gst-libs/gst/rtsp/gstrtspmessage.h:
	  * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	  (parse_npt_range), (parse_clock_range), (parse_smpte_range),
	  (gst_rtsp_range_parse):
	  * gst-libs/gst/rtsp/gstrtsprange.h:
	  * gst-libs/gst/rtsp/gstrtsptransport.c:
	  * gst-libs/gst/rtsp/gstrtspurl.c:
	  Fix some more RTSP docs.
	  Add some missing methods for dealing with messages.

2007-07-24 19:19:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Added beginnings of RTSP documentation.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
	  * gst-libs/gst/rtsp/gstrtspbase64.h:
	  * gst-libs/gst/rtsp/gstrtspconnection.c:
	  (gst_rtsp_connection_connect), (add_auth_header),
	  (gst_rtsp_connection_write), (gst_rtsp_connection_send),
	  (read_body), (gst_rtsp_connection_receive),
	  (gst_rtsp_connection_next_timeout),
	  (gst_rtsp_connection_reset_timeout),
	  (gst_rtsp_connection_set_auth):
	  * gst-libs/gst/rtsp/gstrtspconnection.h:
	  * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
	  * gst-libs/gst/rtsp/gstrtspdefs.h:
	  * gst-libs/gst/rtsp/gstrtspmessage.h:
	  * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	  (parse_npt_range), (parse_clock_range), (parse_smpte_range),
	  (gst_rtsp_range_parse):
	  * gst-libs/gst/rtsp/gstrtspurl.h:
	  Added beginnings of RTSP documentation.

2007-07-24 17:37:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Document the SDP library.
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/sdp/gstsdp.h:
	  * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
	  (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
	  (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
	  (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
	  (gst_sdp_message_get_attribute_val),
	  (gst_sdp_message_add_attribute), (gst_sdp_media_new),
	  (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
	  (gst_sdp_media_get_media), (gst_sdp_media_set_media),
	  (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
	  (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
	  (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
	  (gst_sdp_media_get_format), (gst_sdp_media_add_format),
	  (gst_sdp_media_get_information), (gst_sdp_media_set_information),
	  (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
	  (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
	  (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
	  (gst_sdp_media_set_key), (gst_sdp_media_get_key),
	  (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
	  (gst_sdp_media_get_attribute_val_n),
	  (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
	  (print_media), (gst_sdp_message_dump):
	  * gst-libs/gst/sdp/gstsdpmessage.h:
	  Document the SDP library.
	  Add some of the missing SDPMedia methods.

2007-07-24 11:52:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Move SDP and RTSP from helper objects in -good to a reusable library.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst-libs/gst/Makefile.am:
	  * gst-libs/gst/rtsp/Makefile.am:
	  * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
	  * gst-libs/gst/rtsp/gstrtspbase64.h:
	  * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
	  (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
	  (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
	  (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
	  (parse_response_status), (parse_request_line), (parse_line),
	  (gst_rtsp_connection_read), (read_body),
	  (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
	  (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
	  (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
	  (gst_rtsp_connection_set_auth):
	  * gst-libs/gst/rtsp/gstrtspconnection.h:
	  * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
	  (gst_rtsp_strresult), (gst_rtsp_method_as_text),
	  (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
	  (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
	  (gst_rtsp_find_method):
	  * gst-libs/gst/rtsp/gstrtspdefs.h:
	  * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
	  (gst_rtsp_message_new), (gst_rtsp_message_init),
	  (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
	  (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
	  (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
	  (gst_rtsp_message_free), (gst_rtsp_message_add_header),
	  (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
	  (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
	  (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
	  (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
	  (gst_rtsp_message_dump):
	  * gst-libs/gst/rtsp/gstrtspmessage.h:
	  * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
	  (parse_npt_range), (parse_clock_range), (parse_smpte_range),
	  (gst_rtsp_range_parse), (gst_rtsp_range_free):
	  * gst-libs/gst/rtsp/gstrtsprange.h:
	  * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
	  (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
	  (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
	  (range_as_text), (rtsp_transport_mode_as_text),
	  (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
	  (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
	  (gst_rtsp_transport_free):
	  * gst-libs/gst/rtsp/gstrtsptransport.h:
	  * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
	  (gst_rtsp_url_free), (gst_rtsp_url_set_port),
	  (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
	  * gst-libs/gst/rtsp/gstrtspurl.h:
	  * gst-libs/gst/sdp/Makefile.am:
	  * gst-libs/gst/sdp/gstsdp.h:
	  * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
	  (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
	  (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
	  (gst_sdp_attribute_init), (gst_sdp_message_new),
	  (gst_sdp_message_init), (gst_sdp_message_uninit),
	  (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
	  (gst_sdp_media_uninit), (gst_sdp_media_free),
	  (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
	  (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
	  (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
	  (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
	  (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
	  (gst_sdp_message_get_attribute_val),
	  (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
	  (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
	  (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
	  (gst_sdp_media_get_attribute_val_n),
	  (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
	  (read_string), (read_string_del), (gst_sdp_parse_line),
	  (gst_sdp_message_parse_buffer), (print_media),
	  (gst_sdp_message_dump):
	  * gst-libs/gst/sdp/gstsdpmessage.h:
	  * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	  Move SDP and RTSP from helper objects in -good to a reusable library.
	  Use a proper gst_ namespace.

2007-07-23 18:42:22 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
	  (vorbis_dec_flush_decode):
	  Use the new buffer clipping function from gstaudio here.

2007-07-23 18:26:09 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  API: Add buffer clipping function for raw audio buffers. Fixes #456656.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
	  * gst-libs/gst/audio/audio.h:
	  * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
	  API: Add buffer clipping function for raw audio buffers. Fixes #456656.
	  Also add deprecation guards for gst_audio_structure_set_int() to the
	  header.

2007-07-23 14:45:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Cleanup the docs.

2007-07-23 11:18:35 +0000  Dan Williams <dcbw@redhat.com>

	  gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
	  Original commit message from CVS:
	  Patch by: Dan Williams <dcbw at redhat dot com>
	  * gst/playback/gstplaybasebin.c:
	  (gst_play_base_bin_get_streaminfo_value_array):
	  Don't return NULL when querying the stream info value array but instead
	  return an empty array. Fixes #459204.

2007-07-23 10:41:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gsturidecodebin.c: Init debug category before using it.
	  Original commit message from CVS:
	  * gst/playback/gsturidecodebin.c:
	  Init debug category before using it.

2007-07-21 09:56:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/mixer.h:
	  Add padding vars in place of the signal pointers
	  when building with DISABLE_DEPRECATED so that the
	  interface structure doesn't change size.

2007-07-21 09:21:12 +0000  Marc-Andre Lureau <marcandre.lureau@gmail.com>

	  Fixes: #152864
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * ext/alsa/gstalsamixer.c:
	  * ext/alsa/gstalsamixer.h:
	  * ext/alsa/gstalsamixerelement.c:
	  * ext/alsa/gstalsamixertrack.c:
	  * gst-libs/gst/interfaces/mixer.c:
	  * gst-libs/gst/interfaces/mixer.h:
	  * gst-libs/gst/interfaces/mixeroptions.c:
	  * gst-libs/gst/interfaces/mixeroptions.h:
	  * gst-libs/gst/interfaces/mixertrack.c:
	  * gst-libs/gst/interfaces/mixertrack.h:
	  * tests/check/Makefile.am:
	  * tests/check/libs/mixer.c:
	  Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
	  Fixes: #152864
	  Add support for notifying mixer changes on the message bus, and
	  implement it in alsamixer.
	  API: gst_mixer_get_mixer_flags
	  API: gst_mixer_message_parse_mute_toggled
	  API: gst_mixer_message_parse_record_toggled
	  API: gst_mixer_message_parse_volume_changed
	  API: gst_mixer_message_parse_option_changed
	  API: GstMixerMessageType
	  API: GstMixerFlags

2007-07-20 16:09:03 +0000  Michael Smith <msmith@xiph.org>

	  sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	  (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
	  xcontext->im_format is only for testing XShm support (as the header
	  file comments document). Use xvimage->im_format for everything else.
	  Avoids spurious warnings on buffer allocation before setcaps.

2007-07-20 07:22:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/: We should use $(LIBM).
	  Original commit message from CVS:
	  * tests/examples/volume/Makefile.am:
	  * tests/icles/Makefile.am:
	  We should use $(LIBM).

2007-07-20 06:13:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/icles/Makefile.am: This needs -lm.
	  Original commit message from CVS:
	  * tests/icles/Makefile.am:
	  This needs -lm.

2007-07-18 07:35:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add stdlib include (free, atoi, exit).
	  Original commit message from CVS:
	  * examples/app/appsrc_ex.c:
	  * examples/switch/switcher.c:
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/timidity/gstwildmidi.c:
	  * ext/x264/gstx264enc.c:
	  * gst/mve/mveaudioenc.c: (mve_compress_audio):
	  * gst/rtpmanager/gstrtpclient.c:
	  * gst/rtpmanager/gstrtpjitterbuffer.c:
	  * gst/spectrum/demo-audiotest.c:
	  * gst/spectrum/demo-osssrc.c:
	  * sys/dvb/gstdvbsrc.c:
	  Add stdlib include (free, atoi, exit).

2007-07-16 10:10:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  (gst_basertppayload_class_init), (gst_basertppayload_init),
	  (gst_basertppayload_set_property),
	  (gst_basertppayload_get_property):
	  Don't break ABI, restore previous ranges. Keep the default random
	  selection of timestamp and seqnum offset but as soon as the app sets a
	  specific value, use that one.

2007-07-14 18:33:15 +0000  Bastien Nocera <hadess@hadess.net>

	  sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
	  Original commit message from CVS:
	  Patch by: Bastien Nocera <hadess at hadess dot net>
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	  (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	  (gst_xvimagesink_init), (gst_xvimagesink_class_init):
	  * sys/xvimage/xvimagesink.h:
	  Add option to turn off double-buffering for debugging purposes.
	  Fixes #437169.

2007-07-14 18:20:41 +0000  Jorn Baayen <jorn@openedhand.com>

	  sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
	  Original commit message from CVS:
	  Patch by: Jorn Baayen <jorn at openedhand dot com>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
	  (gst_ximagesink_set_property), (gst_ximagesink_get_property),
	  (gst_ximagesink_init), (gst_ximagesink_class_init):
	  * sys/ximage/ximagesink.h:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
	  (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	  (gst_xvimagesink_init), (gst_xvimagesink_class_init):
	  * sys/xvimage/xvimagesink.h:
	  add 'handle-expose' property. Useful for video widgets which may want to
	  be in control of Expose behaviour. Fixes #380625

2007-07-14 17:23:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  (gst_basertppayload_class_init), (gst_basertppayload_init),
	  (gst_basertppayload_event), (gst_basertppayload_push),
	  (gst_basertppayload_set_property),
	  (gst_basertppayload_get_property),
	  (gst_basertppayload_change_state):
	  * gst-libs/gst/rtp/gstbasertppayload.h:
	  Fix ranges of rtp payloader properties so that the full range can be
	  used in addition to -1 (random).
	  Fix wrong seqnum reporting in caps.
	  Fixes #420326.

2007-07-13 18:12:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videorate/gstvideorate.c: Use boilerplate.
	  Original commit message from CVS:
	  * gst/videorate/gstvideorate.c: (gst_video_rate_init),
	  (gst_video_rate_query):
	  Use boilerplate.
	  Add latency query, might not be perfect yet but already works a lot
	  better. Fixes #442557.

2007-07-13 16:05:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	  (gst_xvimagesink_setcaps):
	  * sys/xvimage/xvimagesink.h:
	  After a caps change, redraw our borders to avoid garbage left there
	  when the image format changes to a smaller size, like 16:9 -> 4:3
	  Also, hold the flow_lock a bit longer in the set_caps while we're
	  fiddling with the xcontext.

2007-07-13 16:02:23 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
	  Original commit message from CVS:
	  * Makefile.am:
	  * configure.ac:
	  * tests/Makefile.am:
	  Remove bogus check for libcheck, since we check for
	  gstreamer-check and it pulls in the required info from there, and we
	  weren't actually _using_ the information for libcheck ourselves
	  anyway.

2007-07-13 15:52:02 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_caps_to_pixfmt):
	  Fix the r_mask test for RGBA32 on little-endian.
	  Fix a stupid typo that would have obviously broken
	  compilation on big-endian, if anyone was testing.

2007-07-12 15:02:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
	  Original commit message from CVS:
	  * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
	  (paint_hline_str4):
	  * gst/videotestsrc/videotestsrc.h:
	  Add alpha to the color struct.
	  Use a default alpha value of 255 instead of 128.

2007-07-12 12:01:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (no_more_pads_full),
	  (setup_source):
	  Clear the dynamic pads counter when starting a new uri. This makes
	  reusing playbin work again.
	  Fixes #454264.

2007-07-12 11:13:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Use pkg-config to locate check.
	  Original commit message from CVS:
	  * configure.ac:
	  Use pkg-config to locate check.

2007-07-11 23:12:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Fix 'make check' build against core CVS.
	  Original commit message from CVS:
	  * configure.ac:
	  * tests/check/elements/volume.c: (GST_START_TEST):
	  Fix 'make check' build against core CVS.

2007-07-10 20:46:41 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/: Make gtk-doc happy.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/propertyprobe.c:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  Make gtk-doc happy.

2007-07-08 13:07:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_callback):
	  Quick hack to make audiosinks stop at EOS when operating in
	  pull-mode; needs to be fixed properly some day.

2007-07-06 18:19:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Fix location of includes in the docs.

2007-07-06 11:40:45 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/avcodec.h:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
	  (gst_ffmpegcsp_avpicture_fill):
	  * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
	  (img_get_alpha_info):
	  Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
	  of the existing BGRA32 and RGBA32 formats with the alpha at the other
	  end of the word. Partially fixes #451908

2007-07-05 08:43:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/: Simplify --extra-dir as gtkdoc scans recursively.
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  * docs/plugins/Makefile.am:
	  Simplify --extra-dir as gtkdoc scans recursively.

2007-07-03 11:52:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
	  (gst_adder_request_new_pad):
	  Make getcaps more robust by not using the proxycaps function. This makes
	  sure that we don't end up recursively calling getcaps upstream.
	  See #316248.

2007-06-29 17:21:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c:
	  Include math.h to fix compilation.

2007-06-29 14:47:42 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
	  Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
	  format, as produced by some dc1394 cameras like the iSight.
	  See http://www.fourcc.org/yuv.php#IYU1

2007-06-28 20:37:58 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
	  Original commit message from CVS:
	  * gst/audioconvert/Makefile.am:
	  * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
	  (check_default), (audio_convert_prepare_context),
	  (audio_convert_clean_context), (audio_convert_convert):
	  * gst/audioconvert/audioconvert.h:
	  * gst/audioconvert/gstaudioconvert.c:
	  (gst_audio_convert_dithering_get_type),
	  (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
	  (gst_audio_convert_init), (gst_audio_convert_set_caps),
	  (gst_audio_convert_set_property), (gst_audio_convert_get_property):
	  * gst/audioconvert/gstaudioconvert.h:
	  * gst/audioconvert/gstaudioquantize.c:
	  (gst_audio_quantize_setup_noise_shaping),
	  (gst_audio_quantize_free_noise_shaping),
	  (gst_audio_quantize_setup_dither),
	  (gst_audio_quantize_free_dither),
	  (gst_audio_quantize_setup_quantize_func),
	  (gst_audio_quantize_setup), (gst_audio_quantize_free):
	  * gst/audioconvert/gstaudioquantize.h:
	  Implement dithering and noise shaping in audioconvert. By default now
	  TPDF dithering (and no noise shaping) will be used when converting
	  from a higher bit depth to 20 bit depth or smaller, otherwise
	  everything will be as it is now.
	  For the last audioconvert in a pipeline it would make sense to
	  use some kind of noise shaping, enabling it by default for all
	  conversions would give undesired results though. Fixes #360246.
	  * tests/check/elements/audioconvert.c: (setup_audioconvert),
	  (GST_START_TEST):
	  Adjust unit test for the new audioconvert.

2007-06-28 11:06:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
	  Use other metrics as well when estimating the buffer level.

2007-06-28 10:21:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Small debug improvement.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
	  Small debug improvement.
	  * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
	  (plugin_init):
	  Tweak the rate estimation period.
	  When calculating the buffer filledness in rate estimation mode, don't
	  mix it with other metrics.

2007-06-28 09:46:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
	  (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
	  When creating the groups, allow for a 5 second, unlimited buffers
	  preroll phase after which we expose the group.
	  When the group is exposed, use a small number of buffers up to a 2
	  second limit. Also disconnect the overrun signal from multiqueue when we
	  exposed the group because it is not needed anymore.

2007-06-27 22:30:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
	  to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
	  (#451707); also, output some debugging info when dealing with
	  freeform strings.
	  * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
	  Add unit test for the above.

2007-06-27 12:55:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
	  Original commit message from CVS:
	  * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
	  Add description for Windows Media RTP caps.
	  * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
	  Remove RTP fields that don't define the format from caps.

2007-06-27 10:14:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
	  Skip empty buffers, but not empty header buffers. That way the original
	  vorbisdec unit test still passes (#451145); also, take into account
	  that those empty packets might carry a granulepos.
	  * tests/check/Makefile.am:
	  * tests/check/elements/vorbisdec.c:
	  (_create_codebook_header_buffer), (_create_audio_buffer),
	  (GST_START_TEST), (vorbisdec_suite):
	  Add unit test that sends an empty packet.

2007-06-27 09:49:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
	  Don't error out on 0-sized packets, just emit a warning because this is
	  not a fatal error. Fixes #451145.

2007-06-25 12:43:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/plugins/: Update docs with caps info.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-base-plugins.args:
	  * docs/plugins/gst-plugins-base-plugins.signals:
	  * docs/plugins/inspect/plugin-adder.xml:
	  * docs/plugins/inspect/plugin-alsa.xml:
	  * docs/plugins/inspect/plugin-audioconvert.xml:
	  * docs/plugins/inspect/plugin-audiorate.xml:
	  * docs/plugins/inspect/plugin-audioresample.xml:
	  * docs/plugins/inspect/plugin-audiotestsrc.xml:
	  * docs/plugins/inspect/plugin-cdparanoia.xml:
	  * docs/plugins/inspect/plugin-decodebin.xml:
	  * docs/plugins/inspect/plugin-decodebin2.xml:
	  * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	  * docs/plugins/inspect/plugin-gdp.xml:
	  * docs/plugins/inspect/plugin-gnomevfs.xml:
	  * docs/plugins/inspect/plugin-libvisual.xml:
	  * docs/plugins/inspect/plugin-ogg.xml:
	  * docs/plugins/inspect/plugin-pango.xml:
	  * docs/plugins/inspect/plugin-playbin.xml:
	  * docs/plugins/inspect/plugin-subparse.xml:
	  * docs/plugins/inspect/plugin-tcp.xml:
	  * docs/plugins/inspect/plugin-theora.xml:
	  * docs/plugins/inspect/plugin-typefindfunctions.xml:
	  * docs/plugins/inspect/plugin-video4linux.xml:
	  * docs/plugins/inspect/plugin-videorate.xml:
	  * docs/plugins/inspect/plugin-videoscale.xml:
	  * docs/plugins/inspect/plugin-videotestsrc.xml:
	  * docs/plugins/inspect/plugin-volume.xml:
	  * docs/plugins/inspect/plugin-vorbis.xml:
	  * docs/plugins/inspect/plugin-ximagesink.xml:
	  * docs/plugins/inspect/plugin-xvimagesink.xml:
	  Update docs with caps info.

2007-06-25 12:04:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.in: Add more files with translatable strings (#450875).
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  Add more files with translatable strings (#450875).

2007-06-23 14:44:07 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
	  The chain should be freed if we error out here, else it will leak.
	  * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
	  (cleanup_decodebin):
	  Don't forget to *properly* remove the signals, else it will leak.

2007-06-22 14:25:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  MAINTAINERS: Updating all the maintainers files
	  Original commit message from CVS:
	  * MAINTAINERS:
	  Updating all the maintainers files

2007-06-21 08:34:46 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
	  (main):
	  Destroy and recreate parse-launch based pipeline after stop to be able
	  to play again. Reorder some code and add more comments.

2007-06-20 11:09:03 +0000  Wim Taymans <wim@fluendo.com>

	  gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (analyze_new_pad):
	  When handling a delayed-caps notification case, mark
	  the group as dynamic so that the nbdynamic count is
	  incremented and decremented correctly. Fixes: #449156
	  Patch by: Wim Taymans <wim@fluendo.com>

2007-06-19 19:13:04 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* win32/common/config.h:
	  gst-libs/gst/audio/gstbaseaudiosink.c
	  Original commit message from CVS:
	  2007-06-19  Andy Wingo  <wingo@pobox.com>
	  * gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_init): Enable pull-mode operation.

2007-06-19 09:34:35 +0000  Michael Smith <msmith@xiph.org>

	  gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Change minimum rate back to 1000 to allow low-sample-rate wav files
	  to play back.

2007-06-17 17:27:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/vi.po: Update translations.
	  Original commit message from CVS:
	  * po/vi.po:
	  Update translations.

2007-06-16 03:42:14 +0000  David Schleef <ds@schleef.org>

	  gst/playback/gstqueue2.c: Fix compile error from ignored return value.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c:
	  Fix compile error from ignored return value.

2007-06-15 15:23:36 +0000  Michael Smith <msmith@xiph.org>

	  gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
	  Original commit message from CVS:
	  * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
	  Update tmpbuf for all neccesary rows, not just one, as is required
	  when downscaling.
	  Fixes #402076.

2007-06-15 11:15:28 +0000  Michael Smith <msmith@xiph.org>

	  tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
	  Original commit message from CVS:
	  * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
	  (eos_buffer_probe):
	  Add a test that ensures we set DELTA_UNIT on all non-header,
	  non-video buffers, if we have a video stream.
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
	  (gst_ogg_mux_process_best_pad):
	  Move setting delta_pad to earlier, where we inspect all pads, so
	  that leading audio pages don't get DELTA_UNIT unset if they come
	  before the first DELTA_UNIT from video pages. Fixes the newly-added
	  test. Fixes #385527.

2007-06-14 19:53:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
	  Original commit message from CVS:
	  * tests/check/pipelines/streamheader.c: (streamheader_suite):
	  Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
	  fails on the p5-ppc64 build bot and the failure looks like it is due
	  to the same issue as #348114, ie. a compiler bug.

2007-06-13 18:20:57 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstqueue2.c: Fix build on MacOSX.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_create_read):
	  Fix build on MacOSX.

2007-06-13 09:01:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
	  Fix compilation on mingw. Fixes #446972.

2007-06-12 08:38:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
	  Original commit message from CVS:
	  Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
	  * gst/playback/gstqueue2.c: (update_buffering),
	  (gst_queue_locked_enqueue):
	  Fix a division by zero when the max percent is <= 0. Fixes #446572.
	  also update the buffering status when receiving events. Fixes #446551.

2007-06-11 11:32:26 +0000  Thiago Sousa Santos <thiagossantos@gmail.com>

	  gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
	  Original commit message from CVS:
	  Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
	  * gst/playback/gstqueue2.c: (gst_queue_peer_query),
	  (gst_queue_handle_src_query):
	  Wait for preroll before attempting to forward a duration query upstream.
	  Fixes #445505.

2007-06-07 21:08:38 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  Use G_GINT64_CONSTANT macro for int64 constant.
	  * win32/common/libgstinterfaces.def:
	  * win32/common/libgsttag.def:
	  Add new exported functions.

2007-06-07 14:25:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
	  The BOS page of the first Dirac video stream needs to come before
	  the BOS page of any Vorbis streams or other audio streams, just like
	  it is with Theora.

2007-06-07 09:11:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Fix compilation.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_get_range):
	  Fix compilation.

2007-06-06 13:36:26 +0000  Thiago Sousa Santos <thiagossantos@gmail.com>

	  gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
	  Original commit message from CVS:
	  Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
	  * gst/playback/gstqueue2.c: (gst_queue_init),
	  (gst_queue_handle_sink_event), (gst_queue_chain),
	  (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
	  (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
	  (gst_queue_src_activate_pull):
	  Add pull based scheduling and fix some deadlocks. Fixes #444523.
	  Does not yet completely work because duration queries upstream won't
	  block yet.

2007-06-06 09:08:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Some more fseeko checks.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/playback/gstqueue2.c: (gst_queue_create_read):
	  Some more fseeko checks.

2007-06-06 08:01:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  configure.ac: check for large file support.
	  Original commit message from CVS:
	  * configure.ac:
	  check for large file support.

2007-06-05 21:36:11 +0000  Sven Arvidsson <sa@whiz.se>

	  gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
	  Original commit message from CVS:
	  Based on a patch by Sven Arvidsson <sa at whiz dot se>:
	  * gst/subparse/gstsubparse.c: (parse_subrip),
	  (subviewer_unescape_newlines), (parse_subviewer),
	  (gst_sub_parse_data_format_autodetect),
	  (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
	  * gst/subparse/gstsubparse.h:
	  Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
	  * tests/check/elements/subparse.c: (GST_START_TEST),
	  (subparse_suite):
	  Add a unit test for both SubViewer formats.

2007-06-05 17:08:04 +0000  Michael Smith <msmith@xiph.org>

	  gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
	  Don't overflow intermediate values when seeking to large time values
	  in audiotestsrc.

2007-06-05 17:02:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Include stdio to define fseeko.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_have_data),
	  (gst_queue_create_read), (gst_queue_read_item_from_file),
	  (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
	  Include stdio to define fseeko.

2007-06-05 16:37:09 +0000  Edward Hervey <edward@fluendo.com>

	  sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
	  Original commit message from CVS:
	  Patch by: Edward Hervey  <edward@fluendo.com>
	  * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
	  (gst_v4lsrc_query):
	  Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.

2007-06-05 16:20:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/Makefile.am:
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
	  Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
	  our own implementation.

2007-06-05 16:19:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_setcaps),
	  (gst_base_rtp_depayload_set_gst_timestamp),
	  (gst_base_rtp_depayload_change_state):
	  Handle timestamp wraparound.

2007-06-05 16:17:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
	  Original commit message from CVS:
	  * gst/playback/gsturidecodebin.c: (no_more_pads_full),
	  (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
	  (gst_uri_decode_bin_change_state):
	  Make sure we name srcpads uniquely even when using different internal
	  decodebins.
	  Signal no-more-pads when no more dynamic elements exist.
	  Remove pads on cleanup.

2007-06-05 16:14:23 +0000  Thiago Sousa Santos <thiagossantos@gmail.com>

	  gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
	  Original commit message from CVS:
	  Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
	  * gst/playback/gstqueue2.c: (gst_queue_class_init),
	  (gst_queue_init), (gst_queue_finalize),
	  (gst_queue_write_buffer_to_file), (gst_queue_have_data),
	  (gst_queue_create_read), (gst_queue_read_item_from_file),
	  (gst_queue_open_temp_location_file),
	  (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
	  (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	  (gst_queue_is_empty), (gst_queue_is_filled),
	  (gst_queue_change_state), (gst_queue_set_temp_location),
	  (gst_queue_set_property):
	  Add support for filebased buffering. Fixes #441264.

2007-06-05 16:05:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
	  (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
	  (caps_notify_group_cb), (gst_decode_group_new),
	  (gst_decode_group_free):
	  Add support for delayed caps fixation when autoplugging.
	  Optimize cases where a multiqueue is not needed/wanted, like right after
	  anything that is not a demuxer.

2007-06-05 16:02:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
	  (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
	  (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
	  consideratly speedup ogg chain detection by not trying to find a base
	  timestamp for skeleton streams.

2007-06-05 16:00:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
	  (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
	  (gst_multi_fd_sink_remove_flush),
	  (gst_multi_fd_sink_remove_client_link),
	  (gst_multi_fd_sink_handle_client_write),
	  (gst_multi_fd_sink_handle_clients):
	  * gst/tcp/gstmultifdsink.h:
	  Add support for remuve_flush.

2007-06-05 15:59:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Add draft design for forcing keyframes in encoders and implement in theoraenc.
	  Original commit message from CVS:
	  * docs/design/draft-keyframe-force.txt:
	  * ext/theora/theoraenc.c: (theora_enc_sink_event),
	  (theora_enc_chain):
	  Add draft design for forcing keyframes in encoders and implement in
	  theoraenc.

2007-06-05 13:22:18 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS

=== release 0.10.13 ===

2007-06-05 12:50:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* gst-plugins-base.doap:
	* win32/common/config.h:
	* win32/vs6/grammar.dsp:
	* win32/vs6/gst_plugins_base.dsw:
	* win32/vs6/libgstadder.dsp:
	* win32/vs6/libgstaudio.dsp:
	* win32/vs6/libgstaudioconvert.dsp:
	* win32/vs6/libgstaudiorate.dsp:
	* win32/vs6/libgstaudioresample.dsp:
	* win32/vs6/libgstaudioscale.dsp:
	* win32/vs6/libgstaudiotestsrc.dsp:
	* win32/vs6/libgstcdda.dsp:
	* win32/vs6/libgstdecodebin.dsp:
	* win32/vs6/libgstdecodebin2.dsp:
	* win32/vs6/libgstdirectsound.dsp:
	* win32/vs6/libgstffmpegcolorspace.dsp:
	* win32/vs6/libgstgdp.dsp:
	* win32/vs6/libgstinterfaces.dsp:
	* win32/vs6/libgstnetbuffer.dsp:
	* win32/vs6/libgstogg.dsp:
	* win32/vs6/libgstpbutils.dsp:
	* win32/vs6/libgstplaybin.dsp:
	* win32/vs6/libgstriff.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstsinesrc.dsp:
	* win32/vs6/libgstsubparse.dsp:
	* win32/vs6/libgsttag.dsp:
	* win32/vs6/libgsttheora.dsp:
	* win32/vs6/libgsttypefindfunctions.dsp:
	* win32/vs6/libgstutils.dsp:
	* win32/vs6/libgstvideo.dsp:
	* win32/vs6/libgstvideorate.dsp:
	* win32/vs6/libgstvideoscale.dsp:
	* win32/vs6/libgstvideotestsrc.dsp:
	* win32/vs6/libgstvolume.dsp:
	* win32/vs6/libgstvorbis.dsp:
	  Release 0.10.13 "What's going on?"
	  Original commit message from CVS:
	  Release 0.10.13 "What's going on?"

2007-06-05 12:32:03 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/de.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2007-05-31 17:08:58 +0000  Wim Taymans <wim@fluendo.com>

	  gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  In riff, the depth is stored in the size field but it just means that
	  the least significant bits are cleared. We can therefore just play
	  the sample as if it had a depth == width. Fixes: #440997
	  Patch by: Wim Taymans <wim@fluendo.com>
	  Patch by: Sebastian Dröge  <slomo@circular-chaos.org>

2007-05-31 16:36:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
	  Original commit message from CVS:
	  * gst-libs/gst/floatcast/floatcast.h:
	  Define inline when needed on win32 builds. Fixes: #441295

2007-05-29 13:38:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (queue_overrun),
	  (no_more_pads_full):
	  Stop buffering when the group is commited because the queues filled up.
	  Fixes #442024.

2007-05-25 10:07:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
	  (gst_alsa_mixer_free), (gst_alsa_mixer_update),
	  (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
	  (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
	  (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
	  * ext/alsa/gstalsamixer.h:
	  * ext/alsa/gstalsamixerelement.c:
	  (gst_alsa_mixer_element_interface_supported),
	  (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
	  (gst_alsa_mixer_element_set_property),
	  (gst_alsa_mixer_element_get_property),
	  (gst_alsa_mixer_element_change_state):
	  * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
	  * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
	  (gst_mixer_option_changed):
	  * gst-libs/gst/interfaces/mixer.h:
	  Revert commits towards #152864 made so far. We'll pick it up again
	  after the 0.10.13 release.

2007-05-24 16:22:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  After an interrupt (PAUSED/flush) assume that the next sample should not
	  be aligned to the previous sample. Fixes #417992.

2007-05-24 15:16:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Don't add channels and rate fields to the template caps for
	  audio/x-dts, as wavparse might not always be able to set them,
	  which would then lead to 'caps are not a real subset of the
	  template caps' warnings.

2007-05-24 11:15:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
	  Handle unknown or invalid pads without crashing, as might occur if
	  a media file like an mp3 is specified as a subtitle file.
	  Fixes: #410039

2007-05-24 10:19:54 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
	  (setup_sinks):
	  Block the subtitle bin output queue before ghosting it and linking,
	  then unblock after. This avoids spurious not-linked errors caused
	  by the queue starting up (because it gets linked when it is ghosted).
	  Fixes: #350299

2007-05-23 15:54:28 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
	  Original commit message from CVS:
	  * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
	  Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
	  file. Avoids flukes where the input gets typefound to some valid but
	  useless type.

2007-05-22 15:45:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
	  (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
	  Add unit test for gnomevfssink seeking and position reporting for
	  file:// URIs.

2007-05-22 15:30:26 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  ext/gnomevfs/gstgnomevfssink.*: see #412648.
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw at skynet be>
	  * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
	  (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
	  (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
	  * ext/gnomevfs/gstgnomevfssink.h:
	  Fix position reporting, especially after a seek (from upstream),
	  see #412648.

2007-05-22 15:04:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
	  Original commit message from CVS:
	  * ext/cdparanoia/gstcdparanoiasrc.c:
	  Repair umlaut.

2007-05-22 11:40:31 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Specify the full valid range for MP3 samplerates. Fixes a regression
	  caused by extra header checks since the last release.

2007-05-21 15:32:42 +0000  Mike Smith <msmith@xiph.org>

	  sys/: Fix a locking-order bug I introduced with my changes the other day.
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	  Fix a locking-order bug I introduced with my changes the other day.
	  Patch by Mike Smith.

2007-05-21 15:24:21 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_handle_data_packet):
	  Don't look inside 0-length packets (which indicate duplicated
	  frames)

2007-05-21 10:25:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Small cleanups.
	  Original commit message from CVS:
	  * ext/cdparanoia/gstcdparanoiasrc.c:
	  (gst_cd_paranoia_src_read_sector):
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_create):
	  Small cleanups.
	  * ext/theora/theoradec.c: (theora_dec_sink_event):
	  Fix typo.
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  Add some FIXME
	  * gst/playback/gstdecodebin.c: (queue_underrun_cb):
	  And some debug info when a FIXME path is hit.

2007-05-21 09:45:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_class_init),
	  (gst_base_rtp_audio_payload_init),
	  (gst_base_rtp_audio_payload_finalize),
	  (gst_base_rtp_audio_payload_handle_frame_based_buffer),
	  (gst_base_rtp_audio_payload_handle_sample_based_buffer),
	  (gst_base_rtp_payload_audio_handle_event):
	  Some cleanups, remove minptime property as it is now in the parent
	  class.
	  Override parent class event function.
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  (gst_basertppayload_class_init), (gst_basertppayload_init),
	  (gst_basertppayload_event), (gst_basertppayload_set_property),
	  (gst_basertppayload_get_property):
	  * gst-libs/gst/rtp/gstbasertppayload.h:
	  Add min-ptime property.
	  Add handle-event vmethod. Fixes #415001.

2007-05-18 17:10:03 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  update spec
	  Original commit message from CVS:
	  update spec

2007-05-18 15:23:43 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/audio/gstbaseaudiosink.c
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_change_state):
	  Fix typo in comment.
	  * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
	  free_dynamics, pad_probe, close_pad_link, try_to_link_1,
	  get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
	  close_link):
	  * gst/playback/gstplaybin.c (gst_play_bin_set_property,
	  gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
	  Remove trailing whitespaces in comments.
	  * gst/volume/Makefile.am:
	  Fix tabs.

2007-05-18 15:10:08 +0000  Marc-Andre Lureau <marcandre.lureau@gmail.com>

	* ChangeLog:
	* gst-libs/gst/interfaces/mixer.h:
	  gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
	  Original commit message from CVS:
	  patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
	  * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
	  set_option, get_option, _gst_reserved):
	  Revert reordering functions (keep ABI).

2007-05-17 17:35:46 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/: When we create our own window, indicate that we handle the
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
	  (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
	  (gst_ximagesink_show_frame):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	  (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
	  (gst_xvimagesink_show_frame):
	  When we create our own window, indicate that we handle the
	  WM_DELETE client message from the window manager, so that it won't
	  kill our window (and our app) along with it. Handle ClientMessage,
	  post an error on the bus, and close the window. Further buffers
	  arriving will result in a FlowError because the window has been
	  destroyed.
	  Fixes: #393975
	  Clean up the X event handling loop and make them the same for
	  both xvimagesink and ximagesink while I'm at it.

2007-05-17 16:27:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
	  Make decodebin2 autoplug depayloaders too.
	  * gst/playback/gsturidecodebin.c: (source_new_pad):
	  Set the newly created decoder in a usable state when autoplugging a
	  dynamic source such as RTSP.

2007-05-17 16:11:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
	  Original commit message from CVS:
	  * gst/playback/gststreaminfo.c: (cb_probe):
	  Ignore video-codec tag for audio streams and ignore audio-codec tags
	  for video streams. Should make codec name collection a bit more
	  robust against sloppy demuxers that send tag events containing both
	  tags down each pad.

2007-05-17 15:22:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (update_rates):
	  Tweak the buffering thresholds a little.
	  Update the buffer size with the previously calculate rate instead of
	  only when we calculate a new rate so that we get smoother buffering
	  updates.
	  * gst/playback/Makefile.am:
	  * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
	  (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
	  (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
	  (gst_uri_decode_bin_get_property), (unknown_type),
	  (add_element_stream), (no_more_pads_full), (no_more_pads),
	  (source_no_more_pads), (new_decoded_pad), (array_has_value),
	  (gen_source_element), (has_all_raw_caps), (analyse_source),
	  (remove_decoders), (make_decoder), (remove_source),
	  (source_new_pad), (setup_source), (decoder_query_init),
	  (decoder_query_duration_fold), (decoder_query_duration_done),
	  (decoder_query_position_fold), (decoder_query_position_done),
	  (decoder_query_latency_fold), (decoder_query_latency_done),
	  (decoder_query_seeking_fold), (decoder_query_seeking_done),
	  (decoder_query_generic_fold), (gst_uri_decode_bin_query),
	  (gst_uri_decode_bin_change_state), (plugin_init):
	  New element that intergrates a source, optional buffering element and
	  decodebin.

2007-05-17 14:17:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
	  Original commit message from CVS:
	  * configure.ac:
	  Bump libtheora requirement to 1.0alpha5 for the pixformat check
	  (also has a .pc file, so we don't need the fallback check any
	  longer). Fixes #438840.

2007-05-17 13:36:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstqueue2.c: fix build.
	  Original commit message from CVS:
	  * gst/playback/gstqueue2.c: (gst_queue_get_type),
	  (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
	  (apply_segment), (apply_buffer), (update_buffering),
	  (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
	  (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	  (gst_queue_handle_sink_event), (gst_queue_is_filled),
	  (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
	  (plugin_init):
	  fix build.

2007-05-17 11:57:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
	  Original commit message from CVS:
	  * gst/playback/Makefile.am:
	  * gst/playback/gstqueue2.c: (gst_queue_get_type),
	  (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
	  (gst_queue_getcaps), (gst_queue_bufferalloc),
	  (gst_queue_acceptcaps), (update_time_level), (apply_segment),
	  (apply_buffer), (update_buffering), (reset_rate_timer),
	  (update_rates), (gst_queue_locked_flush),
	  (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
	  (gst_queue_handle_sink_event), (gst_queue_is_empty),
	  (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
	  (gst_queue_loop), (gst_queue_handle_src_event),
	  (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
	  (gst_queue_src_activate_push), (gst_queue_change_state),
	  (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
	  On our way to playbin2 this is the new network queue that does buffering
	  all by itself using high and low watermarks. It can also measure up and
	  downstream bandwidth to optimally size the queue.

2007-05-17 11:16:14 +0000  Michael Smith <msmith@xiph.org>

	  gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
	  * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
	  Use the segment->last_stop value to calculate the next timestamp to
	  generate after a seek; not the segment->start value.

2007-05-15 20:14:06 +0000  David Schleef <ds@schleef.org>

	  docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled.  This matches the behavior of gtk+.  Fixes #3...
	  Original commit message from CVS:
	  * docs/Makefile.am: Install docs even when --disable-gtk-doc
	  is disabled.  This matches the behavior of gtk+.  Fixes #349099.

2007-05-15 17:11:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
	  Some more chained streaming ogg timestamp fixes.

2007-05-15 16:46:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Add some FIXMEs.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	  (gst_ogg_demux_handle_page):
	  Add some FIXMEs.
	  Fix chain start/stop segment handling based on patch by
	  <ahalda at cs dot mcgill dot ca> see #320984.

2007-05-15 15:33:54 +0000  Michael Smith <msmith@xiph.org>

	  configure.ac: We don't require a C++ compiler. So don't require one.
	  Original commit message from CVS:
	  * configure.ac:
	  We don't require a C++ compiler. So don't require one.

2007-05-15 15:29:17 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	* ChangeLog:
	* ext/alsa/gstalsamixer.c:
	  ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track):
	  Apply some of the cleanup Tim suggested in #152864 afterwards.

2007-05-15 14:01:26 +0000  Marc-Andre Lureau <marcandre.lureau@gmail.com>

	  ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
	  Original commit message from CVS:
	  patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
	  * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
	  _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
	  gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
	  gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
	  gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
	  gst_alsa_mixer_handle_source_callback,
	  gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
	  gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
	  gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
	  gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
	  gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
	  gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
	  * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
	  * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
	  gst_alsa_mixer_element_interface_supported,
	  gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
	  gst_alsa_mixer_element_set_property,
	  gst_alsa_mixer_element_get_property,
	  gst_alsa_mixer_element_change_state):
	  * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
	  * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
	  gst_mixer_option_changed):
	  * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
	  volume_changed, option_changed, _gst_reserved):
	  Implement notification for alsamixer. Fixes #152864

2007-05-15 03:53:11 +0000  David Schleef <ds@schleef.org>

	  gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
	  Original commit message from CVS:
	  * gst/videotestsrc/videotestsrc.c:
	  * gst/videotestsrc/videotestsrc.h:
	  Add support for video/x-raw-bayer.

2007-05-13 01:06:19 +0000  David Schleef <ds@schleef.org>

	  sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c:
	  Add some sanity checking for the XVImage size returned by X.
	  Related to #377400.

2007-05-12 16:18:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_setcaps),
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  Parse and use additional caps fields as described in updated
	  application/x-rtp caps spec.

2007-05-12 16:16:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_collect_chain_info):
	  If there is a stream in a chain without any data packets, ignore the
	  stream in the total length calculations. Might be related to #436820.

2007-05-11 17:33:43 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
	  (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
	  (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
	  (mpeg_video_type_find), (mpeg_video_stream_type_find),
	  (plugin_init):
	  Consolidate and re-work our mpeg system stream detection to probe
	  more packets and produce a higher confidence result. Fixes a
	  regression caused by lowering the typefind probability last year
	  - related to bug #397810. Remove the redundant MPEG-1 specific
	  typefind function, as the new one detects both MPEG-1 & MPEG-2
	  happily.
	  Also cleanup the MPEG elementary and MPEG-TS detection functions a
	  little.
	  Tested against my media test directory, with some improvements and
	  no regressions.

2007-05-10 15:28:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
	  (queue_out_of_data):
	  Connect to the new queue "pushing" signal instead of the broken
	  "running" one.

2007-05-09 21:17:40 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_handle_frame_based_buffer):
	  Move variable declaration before the first instruction.
	  * gst/videotestsrc/videotestsrc.c:
	  Define M_PI if it's not defined yet.
	  * win32/common/libgstrtp.def:
	  Add new exported functions.

2007-05-09 11:54:32 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_handle_type_packet):
	  gst_pad_push_event() does not return a GstFlowReturn!

2007-05-09 11:25:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/: Some small cosmetic changes.
	  Original commit message from CVS:
	  * tests/examples/seek/scrubby.c: (stop_cb), (main):
	  * tests/examples/seek/seek.c: (do_seek):
	  Some small cosmetic changes.

2007-05-08 19:24:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	* ChangeLog:
	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	  gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
	  Original commit message from CVS:
	  * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
	  gst_adder_change_state):
	  * gst/adder/gstadder.h (bps, offset, collect_event, segment,
	  segment_pending, segment_position, segment_rate):
	  Handle playback-rate on adder.

2007-05-07 11:43:31 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
	  Original commit message from CVS:
	  * ext/theora/gsttheoradec.h:
	  * ext/theora/theoradec.c: (gst_theora_dec_reset),
	  (theora_dec_sink_event), (theora_handle_comment_packet),
	  (theora_handle_type_packet), (theora_dec_change_state):
	  Don't push events (newsegment, tags) before initialising the
	  decoder.
	  This is neccesary for seeking to work correctly in gnonlin.

2007-05-04 13:10:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: gst/audiotestsrc/gstaudiotestsrc.c
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst/adder/gstadder.c:
	  * gst/audiotestsrc/gstaudiotestsrc.c
	  (gst_audio_test_src_create_white_noise):
	  * gst/videotestsrc/gstvideotestsrc.c:
	  * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
	  VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
	  volume_sink_template, volume_src_template, gst_volume_init,
	  volume_process_double, volume_process_int16,
	  volume_process_int16_clamp):
	  Doc fixes and formatting.

2007-05-04 12:41:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	  Minimal check for volume's GstController usability; also another
	  test for #422295.

2007-05-04 09:06:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_add_track):
	  Fix it so that it (a) makes sense and (b) doesn't break
	  everything cdda-related including the unit test.

2007-05-04 08:46:59 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_add_track):
	  Fix build when disabling asserts.

2007-05-03 16:29:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
	  When XShm is not available, we might get row strides that are not
	  rounded up to multiples of four; this is bad, because virtually
	  every RGB-processing element in GStreamer assumes rowstrides are
	  rounded up to multiples of four, so let's allocate at least enough
	  memory to avoid crashes in this case. The image will still be
	  displayed distorted though if this happens, so that still needs
	  fixing (maybe by allocating a bigger image with an 'even' width
	  and then clipping it appropriately when rendering - something for
	  Xlib aficionados in any case).

2007-05-03 13:16:21 +0000  Michael Smith <msmith@xiph.org>

	  gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  If a buffer doesn't have a timestamp, assume it's contiguous with
	  the previous buffer, and synthesise timestamps appropriately.

2007-05-03 11:24:00 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
	  Original commit message from CVS:
	  * tests/check/elements/videorate.c: (GST_START_TEST):
	  Set buffer timestamp to a valid value in order to test the buffer
	  really does stay in videorate.

2007-05-03 10:47:22 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
	  Original commit message from CVS:
	  * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	  There is no sensible way to handle incoming buffers which don't have a
	  valid timestamp. We therefore discard them and wait for the next one.

2007-05-01 18:45:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Better error message for text files.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
	  * gst/playback/gstdecodebin2.c: (plugin_init):
	  Better error message for text files.

2007-04-29 14:38:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
	  Fix offset bug in generation RR packets.

2007-04-27 15:33:46 +0000  Julien Moutte <julien@moutte.net>

	  ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
	  Original commit message from CVS:
	  2007-04-27  Julien MOUTTE  <julien@moutte.net>
	  * ext/theora/theoradec.c: (_theora_granule_time),
	  (theora_dec_push_forward), (theora_handle_data_packet),
	  (theora_dec_decode_buffer): Calculate buffer duration correctly
	  to generate a perfect stream (#433888).
	  * gst/audioresample/gstaudioresample.c:
	  (audioresample_check_discont): Glib provides ABS.

2007-04-27 15:01:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
	  (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
	  (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
	  (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	  (gst_rtcp_packet_bye_set_reason):
	  * gst-libs/gst/rtp/gstrtcpbuffer.h:
	  Fix RB block parsing and writing.
	  Add support for constructing BYE packets.

2007-04-25 08:54:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  When posting a warning message because samples were dropped, post something more intelligible than he default error m...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
	  (gst_base_audio_src_create):
	  * po/POTFILES.in:
	  When posting a warning message because samples were dropped, post
	  something more intelligible than he default error message for clock
	  errors which is just confusing in this context (#432984).

2007-04-25 08:10:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
	  (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
	  (read_packet_header), (gst_rtcp_packet_move_to_next),
	  (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
	  (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
	  (gst_rtcp_packet_sdes_get_item_count),
	  (gst_rtcp_packet_sdes_first_item),
	  (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
	  (gst_rtcp_packet_sdes_first_entry),
	  (gst_rtcp_packet_sdes_next_entry),
	  (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
	  (gst_rtcp_packet_sdes_add_entry):
	  * gst-libs/gst/rtp/gstrtcpbuffer.h:
	  Implement code to write SR, RR and SDES packets.

2007-04-24 20:45:24 +0000  Christian Kirbach <Christian.Kirbach@googlemail.com>

	  sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
	  Original commit message from CVS:
	  Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
	  * sys/ximage/ximagesink.c:
	  Fix build if XShm is not available (#432362).

2007-04-24 18:58:25 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
	  Initalize the AudioConvertCtx with zeroes, otherwise it will contain
	  pointers to random memory which are passed to g_free() when
	  audio_convert_prepare_context() is called the first time.

2007-04-24 15:00:07 +0000  Dan Williams <dcbw@redhat.com>

	  gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
	  Original commit message from CVS:
	  Patch by: Dan Williams <dcbw redhat com>
	  * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	  Don't leak incoming buffer if gst_pad_push() returns a
	  non-OK flow. Fixes #432755.
	  * tests/check/elements/videorate.c: (GST_START_TEST),
	  (videorate_suite):
	  Unit test for the above by Yours Truly.

2007-04-23 20:04:28 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
	  (gst_adder_sink_event), (gst_adder_collected):
	  Fix non-flushing segmented seeks, Fixes #340060 for me

2007-04-21 15:29:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: add API keyword
	  Original commit message from CVS:
	  ChangeLog surgery: add API keyword

2007-04-21 15:25:22 +0000  Olivier Crete <tester@tester.ca>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
	  Original commit message from CVS:
	  Patch by: Olivier Crete  <tester at tester ca>
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_class_init),
	  (gst_base_rtp_audio_payload_init),
	  (gst_base_rtp_audio_payload_dispose):
	  Chain up to parent class in dispose function; get rid of
	  unnecessary 'diposed' flag in private structure (#415001).

2007-04-21 15:10:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Some minor docs fixes and additions; also add missing 'Since' bits.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs.types:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_class_init):
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  Some minor docs fixes and additions; also add missing 'Since' bits.

2007-04-21 14:40:45 +0000  Zeeshan Ali <zeenix@gmail.com>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
	  Original commit message from CVS:
	  Patch by: Zeeshan Ali  <zeenix gmail com>
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_handle_frame_based_buffer),
	  (gst_base_rtp_audio_payload_handle_sample_based_buffer),
	  (gst_base_rtp_audio_payload_push):
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  The recently-added gst_base_rtp_audio_payload_push() should take an
	  object of type GstBaseRTPAudioPayload as first argument (#431672).

2007-04-21 14:14:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c:
	  Make more functions static, just because we can.

2007-04-21 13:54:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
	  Original commit message from CVS:
	  * tests/check/elements/audioresample.c:
	  Add unit test for audioresample shutdown crasher (#420106).

2007-04-20 10:42:24 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/subparse/: Use GST_DISABLE_XML here
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c:
	  * gst/subparse/samiparse.c:
	  Use GST_DISABLE_XML here
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
	  (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
	  (gst_xvimagesink_buffer_alloc),
	  (gst_xvimagesink_navigation_send_event):
	  * sys/xvimage/xvimagesink.h:
	  Include stdlib.h when using atoi.
	  * tests/check/elements/playbin.c: (playbin_suite):
	  Use GST_DISABLE_REGISTRY here

2007-04-19 16:58:53 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
	  Original commit message from CVS:
	  * ext/theora/gsttheoraenc.h:
	  * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
	  (theora_enc_sink_event), (theora_enc_change_state):
	  Track initialisation state; don't try to use encoder state if we're
	  not initialised (it'll segfault).

2007-04-18 11:06:42 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/pipelines/.cvsignore: Fix build.
	  Original commit message from CVS:
	  * tests/check/pipelines/.cvsignore:
	  Fix build.

2007-04-17 10:56:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
	  Original commit message from CVS:
	  * gst/app/Makefile.am:
	  Fix CFLAGS and hopefully #430594.

2007-04-17 02:53:16 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Allow random depths between 1 and 32 instead of only multiplies of 8.

2007-04-17 02:04:21 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Set the maximum number of channels for PCM and float in the correct
	  place to have it also used when creating the template caps.

2007-04-17 01:56:07 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Correctly support 4, 6 and 8 channels with normal PCM and float
	  wav files.
	  Fix the depth and signedness calculation in extensible wav files and
	  also handle 1, 2, 4, 6, 8 channels here when a file without channel
	  mask is found.
	  Add support for float, alaw and mulaw in extensible wav files.
	  This allows correct playback of all but 5 files from
	  http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
	  (gst_riff_create_audio_template_caps):
	  Add voxware and float formats to the template caps.

2007-04-16 22:20:03 +0000  Vincent Torri <vtorri@univ-evry.fr>

	  ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
	  Original commit message from CVS:
	  Patch by: Vincent Torri <vtorri at univ-evry dot fr>
	  * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
	  Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
	  Use the correct format strings for integer formats.

2007-04-16 21:44:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-plugins-base.doap:
	  fix release date
	  Original commit message from CVS:
	  fix release date

2007-04-16 21:42:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-plugins-base.doap:
	  fix release date
	  Original commit message from CVS:
	  fix release date

2007-04-15 14:35:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
	  Original commit message from CVS:
	  * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
	  Don't use pad_alloc_buffer_and_set_caps to create a small header
	  packet, or, worse, to create a big temporary video buffer using the
	  src pad.

2007-04-14 12:34:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  Original commit message from CVS:
	  * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
	  * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, buffer_probe_cb, GST_START_TEST):
	  Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.

2007-04-13 22:10:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gstmultifdsink.c:
	  add debug
	  Original commit message from CVS:
	  add debug

2007-04-13 21:55:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* tests/check/pipelines/streamheader.c:
	  tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  Original commit message from CVS:
	  * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
	  GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
	  streamheader_suite):
	  Add another test set up for failure

2007-04-13 21:09:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/gstoggmux.c:
	* gst/gdp/gstgdpdepay.c:
	  debug changes
	  Original commit message from CVS:
	  debug changes

2007-04-13 21:08:11 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
	  GST_START_TEST, streamheader_suite, main):
	  Add a test for the streamheader bug Wim fixed.

2007-04-13 11:42:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/theora/theoradec.c: Fix misleading comment.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_dec_sink_event):
	  Fix misleading comment.

2007-04-13 06:17:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  More sanity checks for the header fields.

2007-04-12 16:36:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  Try encodings from all environment variables, not just those in the
	  first environment variable that is set.

2007-04-12 15:00:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videorate/gstvideorate.c: Add some debug.
	  Original commit message from CVS:
	  * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	  (gst_video_rate_chain):
	  Add some debug.
	  * tests/check/elements/videorate.c: (GST_START_TEST),
	  (videorate_suite):
	  Added check for videorate changing caps handling. Closes #421834.

2007-04-12 12:57:33 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  Use scale functions to avoid overflow when calculating duration of
	  vorbis buffers.

2007-04-12 12:19:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  API: add gst_tag_freeform_string_to_utf8() (#405072).
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/tag/tag.h:
	  * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
	  API: add gst_tag_freeform_string_to_utf8() (#405072).
	  * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
	  Use gst_tag_freeform_string_to_utf8() here.

2007-04-12 10:38:03 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gstmultifdsink.c:
	  log tweaking
	  Original commit message from CVS:
	  log tweaking

2007-04-12 10:03:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
	  Original commit message from CVS:
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
	  (gst_gdp_pay_sink_event):
	  Make sure we set the IN_CAPS flag correctly.
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	  Get the IN_CAPS flag before we call functions that mess with the flags.

2007-04-10 20:37:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* gst/gdp/gstgdppay.c:
	  gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
	  Original commit message from CVS:
	  * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
	  gst_gdp_pay_chain, gst_gdp_pay_sink_event):
	  Only stamp buffers with offset/offset_end right before they get
	  pushed.  This ensures offset continuity, which was not the case
	  before as shown by
	  gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE

2007-04-10 20:25:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/gdp/gstgdpdepay.c:
	* gst/gdp/gstgdppay.c:
	  adding debugging
	  Original commit message from CVS:
	  adding debugging

2007-04-10 11:23:18 +0000  Christian Schaller <uraeus@gnome.org>

	* common:
	* gst-plugins-base.spec.in:
	  update spec file for RTP changes
	  Original commit message from CVS:
	  update spec file for RTP changes

2007-04-06 12:58:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (add_sink),
	  (gst_play_bin_change_state):
	  Activate sync in playbin, we are ready to handle it for live streams.

2007-04-06 09:56:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
	  Original commit message from CVS:
	  * tests/check/elements/playbin.c:
	  (test_sink_usage_video_only_stream), (playbin_suite):
	  Add small test for stream-info-value-array code paths.

2007-04-05 15:44:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_skew_slaving):
	  Don't try to create invalid calibration parameters by making the
	  internal time go backwards, instead make external time go forward.

2007-04-05 10:27:06 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
	  * gst/playback/gstplaybasebin.c: (add_stream):
	  Fix leak in add_stream(), when g_value_set_object() increases the
	  refcount of streaminfo object. Fixes #426250.

2007-04-04 02:45:03 +0000  David Schleef <ds@schleef.org>

	  gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency.  T...
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  * gst/videotestsrc/gstvideotestsrc.h:
	  * gst/videotestsrc/videotestsrc.c:
	  * gst/videotestsrc/videotestsrc.h:
	  Add a test pattern called "circular", which has concentric
	  rings with varying radial frequency.  The main purpose of this
	  pattern is to test fidelity loss in a filter or scaler element.
	  Notably, this pattern is scale invariant, and is optimally viewed
	  with a width (and height) of 400.

2007-04-03 11:10:52 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
	  * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
	  (deactivate_free_recursive):
	  Decodebin2 doesn't unref pads it obtains in some occasions:
	  - multiqueue src pads, when either connecting further or exposing
	  - sink pads of new autoplugged elements
	  - peer pads when recursively freeing elements
	  Fixes #425455.

2007-03-30 17:05:23 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Add audio/x-raw-float support, now that audioconvert support
	  non-native endianness floats.

2007-03-30 15:00:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  gstreamer-plugins-base.pc doesn't exist, it's
	  gstreamer-plugins-base-0.10.pc.

2007-03-29 18:42:34 +0000  René Stadler <mail@renestadler.de>

	  with some minor changes
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler dot de>
	  with some minor changes
	  * gst-libs/gst/floatcast/floatcast.h:
	  Use more efficient float endianness conversion functions that don't
	  involve 2 function calls per value.
	  * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
	  (check_default), (audio_convert_prepare_context):
	  * gst/audioconvert/gstaudioconvert.c:
	  (gst_audio_convert_parse_caps), (make_lossless_changes):
	  Support non-native endianness floats as input and output.
	  Fixes #339838.
	  * tests/check/elements/audioconvert.c: (verify_convert),
	  (GST_START_TEST):
	  Add unit tests for the non-native endianness float conversions.

2007-03-29 16:23:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_base_init),
	  (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
	  (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	  (gst_base_rtp_depayload_set_gst_timestamp),
	  (gst_base_rtp_depayload_change_state),
	  (gst_base_rtp_depayload_set_property),
	  (gst_base_rtp_depayload_get_property):
	  * gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Add Private structure.
	  Bring element code to 2007.
	  Parse clock-base caps param and use it when generating the
	  newsegment.
	  Reset variables before going to PAUSED.
	  Fix some docs.

2007-03-29 16:20:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Add RTCP docs.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_get_adapter):
	  Add RTCP docs.
	  Fix some more docs.
	  * gst-libs/gst/rtp/Makefile.am:
	  * gst-libs/gst/rtp/gstrtcpbuffer.c:
	  (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
	  (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
	  (gst_rtcp_buffer_get_packet_count), (read_packet_header),
	  (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
	  (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
	  (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
	  (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
	  (gst_rtcp_packet_sr_get_sender_info),
	  (gst_rtcp_packet_sr_set_sender_info),
	  (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
	  (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
	  (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
	  (gst_rtcp_packet_sdes_get_chunk_count),
	  (gst_rtcp_packet_sdes_first_chunk),
	  (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
	  (gst_rtcp_packet_sdes_first_item),
	  (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
	  (gst_rtcp_packet_bye_get_ssrc_count),
	  (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
	  (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
	  (gst_rtcp_packet_bye_get_reason_len),
	  (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
	  * gst-libs/gst/rtp/gstrtcpbuffer.h:
	  Add new helper object for parsing and creating RTCP messages.

2007-03-29 12:07:02 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  PCM samples with width=8 must be always unsigned, no matter what
	  depth they have.

2007-03-29 11:24:47 +0000  Andy Wingo <wingo@pobox.com>

	  gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
	  Original commit message from CVS:
	  2007-03-29  Andy Wingo  <wingo@pobox.com>
	  * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
	  perfect offsets also, not just timestamps.
	  * tests/check/elements/videorate.c (test_more): Test that given
	  any incoming offsets, that videorate produces perfect offsets.

2007-03-29 10:19:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-ids.h:
	  Add some more RIFF formats.

2007-03-29 10:17:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtpbuffer.c:
	  (gst_rtp_buffer_default_clock_rate):
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Fix fixed payload names and docs.
	  Added method to get the default clock rates of fixed payload types.
	  API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()

2007-03-28 15:24:40 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
	  Original commit message from CVS:
	  * tests/check/pipelines/.cvsignore:
	  Add new vorbisdec test to cvsignore.

2007-03-28 14:50:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
	  (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
	  (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
	  (gst_base_audio_sink_set_property),
	  (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
	  (clock_convert_external), (gst_base_audio_sink_resample_slaving),
	  (gst_base_audio_sink_skew_slaving),
	  (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
	  (gst_base_audio_sink_async_play):
	  * gst-libs/gst/audio/gstbaseaudiosink.h:
	  Store private stuff in GstBaseAudioSinkPrivate.
	  Add configurable clock slaving modes property.
	  API:: GstBaseAudioSink::slave-method property
	  Some more latency reporting tweaks.
	  Added skew based clock slaving correction and make it the default until
	  the resampling method is more robust.

2007-03-27 12:44:14 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c:
	  Add docs to the integer pack functions and implement proper
	  rounding. Before we had rounding towards negative infinity, i.e.
	  always the smaller number was taken. Now we use natural rounding,
	  i.e. rounding to the nearest integer and to the one with the largest
	  absolute value for X.5. The old rounding introduced some minor
	  distortions. Fixes #420079
	  * tests/check/elements/audioconvert.c: (GST_START_TEST):
	  Fix one unit test that assumed the old rounding and added unit tests
	  for checking signed/unsigned int16 <-> signed/unsigned int16 with
	  depth 8, one for signed int16 <-> unsigned int16 and one for the new
	  rounding from signed int32 to signed/unsigned int16.

2007-03-27 11:31:17 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
	  (gst_audio_convert_transform_caps):
	  Fix typo in debug line introduced recently, as pointed out on irc.

2007-03-27 10:17:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  * tests/check/libs/tag.c: (GST_START_TEST):
	  Make sure we parse floating-point numbers in vorbis comments
	  correctly with either '.' or ',' as separator, no matter what
	  the current locale is. Add unit test for this too.

2007-03-27 09:37:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/pipelines/vorbisdec.c:
	  commit new file
	  Original commit message from CVS:
	  commit new file

2007-03-26 22:38:19 +0000  René Stadler <mail@renestadler.de>

	  gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
	  Original commit message from CVS:
	  Patch by: René Stadler  <mail at renestadler de>
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
	  When writing out floating-point numbers to vorbis comment tags, always
	  use the same character as separator no matter what the current locale is
	  (fixes #423051).
	  * tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit tests for replaygain tags in vorbis comments (closes #423055).

2007-03-26 20:56:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
	  vorbis_handle_data_packet):
	  Correctly set DURATION to generate a timestamp-continuous stream.
	  One bug left at the end; see
	  ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
	  Add a test to check this.  Without the above patch this test fails.

2007-03-26 11:44:07 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/Makefile.am:
	  The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.

2007-03-23 15:43:24 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  update spec file
	  Original commit message from CVS:
	  update spec file

2007-03-23 12:32:33 +0000  Michael Smith <msmith@xiph.org>

	  gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
	  Original commit message from CVS:
	  * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
	  (gst_video_rate_reset), (gst_video_rate_chain):
	  If videorate changes caps, we can no longer use the old buffer
	  (which may have a different size, incompatible with our caps).
	  So don't do that; just duplicate the new frame more times.

2007-03-22 17:43:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
	  Remove playbin's override of the set_clock vmethod. It's irrelevant
	  after Wim's commit on the 19th.

2007-03-22 14:37:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
	  Original commit message from CVS:
	  * gst-libs/gst/app/Makefile.am:
	  Use GST_ALL_LDFLAGS, which actually exists, but maybe David
	  can confirm that was what he wanted.

2007-03-22 09:26:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
	  (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
	  * ext/gnomevfs/gstgnomevfssrc.h:
	  Don't cache file sizes. Fixes #341078.

2007-03-21 11:03:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (add_sink):
	  Use GST_PTR_FORMAT to log caps.

2007-03-21 10:23:11 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha <ganadist at chollian net>
	  * gst/subparse/samiparse.c: (handle_start_font):
	  Special-case some more colour names that pango doesn't handle by
	  default. Fixes #420578.

2007-03-20 11:49:55 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  If we get a zero-sized input buffer, don't pass it to libvorbis, as
	  that marks EOS internally. After that, libvorbis will buffer all
	  input data, and encode none of it, eventually leading to memory
	  exhaustion.

2007-03-19 10:52:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (remove_fakesink):
	  Don't post STATE_DIRTY anymore.
	  * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
	  (gst_play_bin_change_state):
	  Remove stream_time reset in seek handling, core does that now.
	  Disable clocking for live pipelines by forcing a NULL clock to the
	  complete pipeline, core is too smart now for our previous hack.
	  We can always autoplug in PAUSED now.

2007-03-18 03:14:01 +0000  David Schleef <ds@schleef.org>

	  REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
	  Original commit message from CVS:
	  * REQUIREMENTS:  Update this file, change the formatting to make
	  it more consistent, plus more machine readable.

2007-03-16 17:29:09 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	  (strip_width_64), (append_with_other_format):
	  Previous fix was too simplistic, and broke the tests. Use a better
	  approach; only strip 64 from widths for integer audio.

2007-03-16 16:42:23 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	  (gst_audio_convert_transform_caps):
	  We don't support 64 bit integer audio, so don't try to claim we can.
	  Stops us producing caps don't match our template caps.
	  Update comments.

2007-03-15 10:52:21 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c:
	  (audioresample_check_discont), (audioresample_transform):
	  Don't trigger discontinuities for very small imperfections; a filter
	  flush will sound bad, and many plugins have rounding errors leading
	  to these.

2007-03-14 21:11:18 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  Add min-ptime property to RTP base audio payloader. Patch by
	  olivier.crete@collabora.co.uk.
	  Fixes #415001
	  Indentation/whitespace/documentation fixes.

2007-03-14 17:16:30 +0000  Julien Moutte <julien@moutte.net>

	  gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
	  Original commit message from CVS:
	  2007-03-14  Julien MOUTTE  <julien@moutte.net>
	  * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
	  (audioresample_transform_size), (audioresample_do_output),
	  (audioresample_transform), (audioresample_pushthrough): Handle
	  discontinuous streams.
	  * gst/audioresample/gstaudioresample.h:
	  * tests/check/elements/audioresample.c:
	  (test_discont_stream_instance), (GST_START_TEST),
	  (audioresample_suite): Add a test for discontinuous streams.
	  * win32/common/config.h: Updated.

2007-03-14 15:16:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  po/: Update translations from translation project.
	  Original commit message from CVS:
	  * po/af.po:
	  * po/az.po:
	  * po/cs.po:
	  * po/en_GB.po:
	  * po/it.po:
	  * po/nb.po:
	  * po/nl.po:
	  * po/or.po:
	  * po/sq.po:
	  * po/sr.po:
	  * po/sv.po:
	  * po/uk.po:
	  * po/vi.po:
	  Update translations from translation project.

2007-03-14 15:05:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/gdp/gstgdpdepay.c:
	  add buffer logging
	  Original commit message from CVS:
	  add buffer logging

2007-03-14 14:48:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
	  Original commit message from CVS:
	  * gst/audioresample/debug.h:
	  * gst/audioresample/resample.c: (resample_init):
	  Since I really am not interested in a debug line for each sample
	  being processed, move the library's debugging to its own category,
	  libaudioresample

2007-03-14 14:09:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/audioresample/gstaudioresample.c:
	  add debugging and reformat docs
	  Original commit message from CVS:
	  add debugging and reformat docs

2007-03-12 23:29:07 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_handle_type_packet):
	  Since the plugin doesn't support anything other than 4:2:0 right
	  now, post an error and fail if we get something else. Won't matter
	  until libtheora supports the other pixel formats, but hopefully
	  that'll be soon...

2007-03-12 15:50:35 +0000  Alex Lancaster <alexlan@fedoraproject.org>

	* ChangeLog:
	  I'm too lazy to comment this
	  Original commit message from CVS:
	  Mention Patch by: Alex Lancaster in a recent commit.

2007-03-12 11:47:42 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
	  Original commit message from CVS:
	  * examples/app/.cvsignore:
	  The buildbot demands .cvsignore files, and I comply.

2007-03-11 00:48:26 +0000  David Schleef <ds@schleef.org>

	  Add appsrc/appsink example.
	  Original commit message from CVS:
	  * configure.ac:
	  * examples/Makefile.am:
	  * examples/app/Makefile.am:
	  * examples/app/appsrc_ex.c:
	  Add appsrc/appsink example.
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/app/gstapp.c:
	  * gst-libs/gst/app/gstappsink.c:
	  * gst-libs/gst/app/gstappsink.h:
	  * gst/app/gstapp.c:
	  Add appsink.

2007-03-10 15:59:33 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
	  Use gst_guint64_to_gdouble for conversion.
	  * win32/MANIFEST:
	  Add new files to the win32 MANIFEST.
	  * win32/common/libgstaudio.def:
	  * win32/common/libgstpbutils.def:
	  Add new exported functions.
	  * win32/vs6/gst_plugins_base.dsw:
	  * win32/vs6/libgstdecodebin.dsp:
	  * win32/vs6/libgstplaybin.dsp:
	  Change the link to libgstpbutils.lib.
	  * win32/vs6/libgstdecodebin2.dsp:
	  Add a new project for decodebin2.
	  * win32/vs6/libgstpbutils.dsp:
	  Add a new project for pbutils.

2007-03-10 12:18:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Also accept partial dates with only year and month,
	  like 1999-12-00 (fixes #410396 even more).
	  * tests/check/libs/tag.c: (GST_START_TEST):
	  Add unit test for the above.

2007-03-10 11:21:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
	  Original commit message from CVS:
	  * tests/check/elements/subparse.c: (GST_START_TEST),
	  (subparse_suite):
	  Add unit test for MPL2 subtitle format (#413799).

2007-03-10 11:17:52 +0000  Kamil Pawlowski <kamilpe@gmail.com>

	  gst/subparse/: Add support for MPL2 subtitle format (#413799).
	  Original commit message from CVS:
	  Patch by: Kamil Pawlowski  <kamilpe gmail com>
	  * gst/subparse/Makefile.am:
	  * gst/subparse/gstsubparse.c:
	  (gst_sub_parse_data_format_autodetect),
	  (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
	  (gst_subparse_type_find):
	  * gst/subparse/gstsubparse.h:
	  * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
	  * gst/subparse/mpl2parse.h:
	  Add support for MPL2 subtitle format (#413799).

2007-03-09 17:33:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: We require core CVS for the new buffer metadata copy functions.
	  Original commit message from CVS:
	  * configure.ac:
	  We require core CVS for the new buffer metadata copy functions.

2007-03-09 16:51:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c:
	  Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	  Fixes #414496.

2007-03-09 16:46:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libvisual/visual.c: Improve adapter usage and comments.
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
	  (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
	  Improve adapter usage and comments.

2007-03-09 16:38:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Use new metadata copy function.
	  Original commit message from CVS:
	  * ext/pango/gsttextrender.c: (gst_text_render_chain):
	  * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
	  * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
	  Use new metadata copy function.
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_transform):
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
	  Basetransform copied the metadata for us.

2007-03-09 16:28:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
	  (gst_text_overlay_video_event):
	  Some more logging. Only accept newsegment events in TIME format and
	  send a WARNING message if they are not in TIME format.
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	  (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
	  (gst_sub_parse_chain), (gst_sub_parse_sink_event):
	  * gst/subparse/gstsubparse.h:
	  No need to allocate GstSegment structure dynamically, just put it
	  into the instance structure; ignore newsegment events in BYTE
	  format and in particular don't let it overwrite our saved TIME
	  segment from the last seek.

2007-03-09 13:05:04 +0000  Michael Smith <msmith@xiph.org>

	  gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
	  Replace AC3 typefinder with one that isn't terrible, and actually
	  works usefully.

2007-03-09 12:22:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c:
	  (gst_audio_convert_transform):
	  fix error category and translatable string

2007-03-09 11:23:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  pkgconfig/: Fix up utils => pbutils here too.
	  Original commit message from CVS:
	  * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	  * pkgconfig/gstreamer-plugins-base.pc.in:
	  Fix up utils => pbutils here too.

2007-03-09 10:49:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (handle_buffer):
	  Break out of loop in chain function as soon as possible if we get
	  a non-OK flow return.

2007-03-08 18:26:07 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
	  Original commit message from CVS:
	  * tests/check/elements/alsa.c: (GST_START_TEST):
	  Unref the mixer if the state change fails too (if the
	  alsa devices are inaccessible, for example)

2007-03-08 17:49:46 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Don't test libvisual elements in the states check, because libvisual
	  seems to leak internally.
	  Re-enable the alsa and states tests now that there's new suppressions
	  in gst.supp.
	  * tests/check/elements/alsa.c: (GST_START_TEST):
	  Don't leak the alsamixer we instantiated.

2007-03-08 15:22:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
	  (gst_ximagesink_change_state), (gst_ximagesink_reset),
	  (gst_ximagesink_finalize):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
	  (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
	  Move some cleanup stuff from the state change handler into a _reset()
	  function that can be called from _finalize(). This ensures that things
	  get freed even if (for some reason) the NULL->READY state transition
	  fails in the parent class.
	  Even if a parent state change fails, process our downward state change
	  logic instead of bailing out early.
	  Free the correct xcontext pointer in ximagesink's xcontext_clear.

2007-03-08 12:53:51 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/alsa/gstalsasink.c: Extra log line.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_open):
	  Extra log line.
	  * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
	  * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
	  Use pango_font_description_set_family_static instead of
	  pango_font_description_set_family to save a string copy (it was
	  leaking due to the strdup anyway)
	  * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
	  * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
	  * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
	  * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
	  Chain up in finalize.

2007-03-07 18:50:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/mixertrack.c:
	  (gst_mixer_track_class_init), (gst_mixer_track_get_property),
	  (gst_mixer_track_set_property):
	  API: add "untranslated-label" property which should be set by
	  implementations at construct time (#414645).
	  * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
	  * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Set "untranslated-label" when constructing mixer track objects.
	  * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
	  Unit test to check the above.

2007-03-07 17:15:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Fix confusing debug message.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
	  Fix confusing debug message.

2007-03-07 17:12:54 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-plugins-base.doap: update doap file with new version
	  Original commit message from CVS:
	  * gst-plugins-base.doap:
	  update doap file with new version

2007-03-07 17:05:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gstmultifdsink.c:
	  update docs
	  Original commit message from CVS:
	  update docs

2007-03-07 16:56:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS

=== release 0.10.12 ===

2007-03-07 16:46:51 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-decodebin2.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* win32/common/config.h:
	  Release 0.10.12
	  Original commit message from CVS:
	  Release 0.10.12

2007-03-07 15:35:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* common:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/de.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2007-03-06 12:31:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Bump version to 0.10.11.4 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  Bump version to 0.10.11.4 pre-release

2007-03-06 12:10:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_async_play):
	  Fix regression that made GStreamer skip the first samples of audio.
	  Fixes #414684.

2007-03-05 11:21:13 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Bump version to 0.10.11.3 pre-release
	  Original commit message from CVS:
	  * configure.ac:
	  Bump version to 0.10.11.3 pre-release

2007-03-05 09:35:29 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  Update paths for the rename from utils to pbutils to fix the build.

2007-03-05 09:27:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
	  Original commit message from CVS:
	  * gst-libs/gst/pbutils/Makefile.am:
	  Change directory to install headers in from gst/utils to gst/pbutils
	  as well.

2007-03-04 23:41:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/libs/.gitignore:
	  moap ignore
	  Original commit message from CVS:
	  moap ignore

2007-03-04 23:41:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* win32/common/config.h:
	* win32/common/libgstutils.def:
	  update defs
	  Original commit message from CVS:
	  update defs

2007-03-04 23:39:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  rename utils to pbutils
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/Makefile.am:
	  * gst-libs/gst/interfaces/mixer.c:
	  * gst-libs/gst/pbutils/Makefile.am:
	  * gst-libs/gst/pbutils/descriptions.c:
	  (gst_pb_utils_get_source_description),
	  (gst_pb_utils_get_sink_description),
	  (gst_pb_utils_get_decoder_description),
	  (gst_pb_utils_get_encoder_description),
	  (gst_pb_utils_get_element_description),
	  (gst_pb_utils_add_codec_description_to_tag_list),
	  (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
	  * gst-libs/gst/pbutils/descriptions.h:
	  * gst-libs/gst/pbutils/install-plugins.c:
	  * gst-libs/gst/pbutils/install-plugins.h:
	  * gst-libs/gst/pbutils/missing-plugins.c:
	  (gst_missing_uri_source_message_new),
	  (gst_missing_uri_sink_message_new),
	  (gst_missing_element_message_new),
	  (gst_missing_decoder_message_new),
	  (gst_missing_encoder_message_new),
	  (gst_missing_plugin_message_get_description):
	  * gst-libs/gst/pbutils/missing-plugins.h:
	  * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
	  * gst-libs/gst/pbutils/pbutils.h:
	  * gst-libs/gst/utils/Makefile.am:
	  * gst-libs/gst/utils/base-utils.c:
	  * gst-libs/gst/utils/base-utils.h:
	  * gst-libs/gst/utils/descriptions.c:
	  * gst-libs/gst/utils/descriptions.h:
	  * gst-libs/gst/utils/install-plugins.c:
	  * gst-libs/gst/utils/install-plugins.h:
	  * gst-libs/gst/utils/missing-plugins.c:
	  * gst-libs/gst/utils/missing-plugins.h:
	  * gst-plugins-base.spec.in:
	  * gst/playback/Makefile.am:
	  * gst/playback/gstdecodebin.c:
	  * gst/playback/gstdecodebin2.c:
	  * gst/playback/gstplaybasebin.c: (setup_subtitle),
	  (gen_source_element):
	  * gst/playback/gstplaybin.c: (plugin_init):
	  * tests/check/Makefile.am:
	  * tests/check/libs/pbutils.c: (GST_START_TEST),
	  (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
	  * tests/check/libs/utils.c:
	  rename utils to pbutils

2007-03-03 10:23:03 +0000  David Schleef <ds@schleef.org>

	  gst-libs/gst/app/Makefile.am: Install the headers.
	  Original commit message from CVS:
	  * gst-libs/gst/app/Makefile.am:
	  Install the headers.

2007-03-03 10:10:30 +0000  David Schleef <ds@schleef.org>

	  gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
	  Original commit message from CVS:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/app/gstappbuffer.c:
	  * gst-libs/gst/app/gstappbuffer.h:
	  * gst-libs/gst/app/gstappsrc.c:
	  Add GstAppBuffer that includes a callback and closure for
	  proper handling of data chunks.

2007-03-03 09:06:06 +0000  David Schleef <ds@schleef.org>

	  gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
	  Original commit message from CVS:
	  * gst-libs/gst/app/gstappsrc.c:
	  * gst-libs/gst/app/gstappsrc.h:
	  Hacking to address issues in 413418.

2007-03-03 08:16:57 +0000  David Schleef <ds@schleef.org>

	  Move the app library to gst-libs/gst/app (duh!)
	  Original commit message from CVS:
	  * Makefile.am:
	  * configure.ac:
	  * ext/Makefile.am:
	  * gst-libs/gst/Makefile.am:
	  * gst-libs/gst/app/Makefile.am:
	  * gst-libs/gst/app/gstapp.c:
	  * gst-libs/gst/app/gstappsrc.c:
	  * gst-libs/gst/app/gstappsrc.h:
	  * gst/app/Makefile.am:
	  * gst/app/gstapp.c:
	  * gst/app/gstappsrc.c:
	  * gst/app/gstappsrc.h:
	  Move the app library to gst-libs/gst/app (duh!)

2007-03-02 12:59:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Add documentation for decodebin2 that indicates that the API is still unstable.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-decodebin2.xml:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
	  Add documentation for decodebin2 that indicates that the API
	  is still unstable.

2007-03-01 18:50:00 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
	  Original commit message from CVS:
	  * configure.ac:
	  Update to 0.10.11.2 (0.10.12 pre-release)

2007-03-01 17:29:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_async_play):
	  base time is irrelevant here.

2007-03-01 17:01:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/: Improve debugging.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
	  * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
	  Improve debugging.
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_query), (gst_base_audio_sink_event),
	  (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
	  Improve latency and clock slaving calculations.
	  Improve slave clock calibration.
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_commit_full):
	  When we are asked to render N sample to 0 bytes, return N.

2007-03-01 16:48:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasink.*: Remove unused dispose function.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
	  (gst_alsasink_write), (gst_alsasink_reset):
	  * ext/alsa/gstalsasink.h:
	  Remove unused dispose function.
	  Rename lock to not interfere with alsasrc lock.
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
	  (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
	  (gst_alsasrc_read), (gst_alsasrc_reset):
	  * ext/alsa/gstalsasrc.h:
	  Implement finalize function.
	  Use lock to protect alsa access.
	  Implement _reset.
	  Fine tune sw params.

2007-03-01 10:20:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* configure.ac:
	  typo
	  Original commit message from CVS:
	  typo

2007-02-28 19:27:28 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: Convert to new AG_GST style.
	  Original commit message from CVS:
	  * configure.ac:
	  Convert to new AG_GST style.

2007-02-28 15:17:20 +0000  Ed Catmur <ed@catmur.co.uk>

	  gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
	  Original commit message from CVS:
	  Patch by: Ed Catmur <ed at catmur dot co dot uk>
	  * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
	  (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
	  Fix race condition when rapidly switching visualisations in playbin.
	  Fixes #401029.

2007-02-28 15:11:59 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Include local stuff before system installed things in LDFLAGS and
	  CFLAGS.

2007-02-28 15:10:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Improve debugging.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
	  Improve debugging.

2007-02-28 15:05:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/v4l/: Fix duration and timestamping, taking latency into account.
	  Original commit message from CVS:
	  * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
	  (gst_v4lsrc_fixate), (gst_v4lsrc_query):
	  * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
	  Fix duration and timestamping, taking latency into account.
	  Implement latency query.

2007-02-28 15:02:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
	  (gst_audio_clock_new):
	  Fix clock name.
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_init), (gst_base_audio_sink_query):
	  * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
	  (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
	  (gst_base_audio_src_create):
	  Improve latency query code.
	  Use proper clock names.

2007-02-28 12:57:46 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/generic/states.c:
	  plug test leak
	  Original commit message from CVS:
	  plug test leak

2007-02-28 12:44:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/generic/states.c: Copy the states.c test from core again
	  Original commit message from CVS:
	  * tests/check/generic/states.c: (GST_START_TEST):
	  Copy the states.c test from core again
	  * tests/check/Makefile.am:
	  ignore cdio and cdparanoiasrc

2007-02-28 12:08:27 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	  (double_hq), (audio_convert_get_func_index), (check_default),
	  (audio_convert_prepare_context), (audio_convert_convert):
	  Also make valgrind happy and avoid copying data in some cases.

2007-02-28 11:58:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/generic/states.c:
	  use a macro
	  Original commit message from CVS:
	  use a macro

2007-02-28 11:47:45 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	  (double_hq), (audio_convert_get_func_index),
	  (audio_convert_prepare_context), (audio_convert_convert):
	  * gst/audioconvert/gstaudioconvert.c:
	  (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
	  (gst_audio_convert_transform_caps):
	  * tests/check/elements/audioconvert.c: (GST_START_TEST),
	  (audioconvert_suite):
	  Don't run inplace if that overwrites source data as we go. Add more
	  tests. Fixes #339837 even more.

2007-02-27 18:45:37 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
	  Original commit message from CVS:
	  2007-02-27  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
	  (msg_segment_done): Fix various seeking bugs (Slider was not
	  updating when doing a non flushing seek, Reverse playback
	  on segment seek was wrong).

2007-02-26 21:01:03 +0000  David Schleef <ds@schleef.org>

	  Add a new plugin/library to make it easy for apps to shove data into a pipeline.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/app/Makefile.am:
	  * gst/app/gstapp.c:
	  * gst/app/gstappsrc.c:
	  * gst/app/gstappsrc.h:
	  Add a new plugin/library to make it easy for apps to shove
	  data into a pipeline.

2007-02-26 11:48:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (stop_seek):
	  When we stop scrubbing, don't leave the pipeline PLAYING when we
	  requested a PAUSED state.

2007-02-25 23:51:03 +0000  René Stadler <mail@renestadler.de>

	  gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler de>
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Parse date strings in vorbis comments that have an invalid (zero)
	  month or day (#410396).
	  * tests/check/libs/tag.c: (GST_START_TEST):
	  Test case for the above.

2007-02-24 20:12:49 +0000  Loïc Minier <lool+gnome@via.ecp.fr>

	  Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
	  Original commit message from CVS:
	  Patch by: Loïc Minier <lool+gnome at via ecp fr>
	  * configure.ac:
	  * ext/alsa/Makefile.am:
	  * gst/audiotestsrc/Makefile.am:
	  Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).

2007-02-23 18:49:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c:
	  Improve docs: point out that the application needs to assist playbin
	  with buffering.

2007-02-23 13:10:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
	  Original commit message from CVS:
	  * gst-libs/gst/utils/install-plugins.c:
	  * gst-libs/gst/utils/missing-plugins.c:
	  * tests/check/libs/utils.c: (missing_msg_check_getters):
	  Change GStreamer marker prefix in detail string from 'gstreamer.net'
	  to just 'gstreamer'. Document the caps string component of the
	  decoder/encoder detail a bit better, since not everyone will be
	  familiar with the GStreamer media type/caps system (but they better
	  enjoy nested itemized lists).

2007-02-22 12:57:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
	  Original commit message from CVS:
	  * gst-libs/gst/netbuffer/gstnetbuffer.c:
	  (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
	  Fix copying of GstNetBuffer (would crash before, or at least lead to
	  invalid memory access, #410772), for now by copying the GstBuffer copy
	  code from the core over here so we can copy the GstBuffer fields on a
	  provided buffer instance (of type GstNetBuffer in this case). Would be
	  better to fix this with some support by the core though (and in the long
	  run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
	  * tests/check/Makefile.am:
	  Enable unit test for GstNetBuffer.

2007-02-22 11:04:10 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  gst-libs/gst/audio/gstbaseaudiosink.c
	  Original commit message from CVS:
	  2007-02-22  Andy Wingo  <wingo@pobox.com>
	  * gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_init): Disable pull-mode activation until we
	  figure out how to make audio sinks go to PLAYING.

2007-02-22 09:04:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
	  (double_hq), (audio_convert_get_func_index),
	  (audio_convert_prepare_context), (audio_convert_convert):
	  * gst/audioconvert/audioconvert.h:
	  * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
	  (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
	  * gst/audioconvert/gstchannelmix.h:
	  * tests/check/elements/audioconvert.c: (GST_START_TEST):
	  Add float as an intermediate format, as well as float mixing. Enable
	  test that was failing before. Fixes #339837

2007-02-21 16:12:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (do_seek):
	  Undo the previous commit: -1 as a stop time implies that the stop
	  time is the end of file, clearing any previously configured segment.

2007-02-21 15:36:26 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (do_seek):
	  Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.

2007-02-21 13:55:54 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/volume/gstvolume.c: Unbreak volume, value remains gint.
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c: (volume_process_int16),
	  (volume_process_int16_clamp), (volume_set_caps):
	  Unbreak volume, value remains gint.

2007-02-21 13:08:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c: (volume_choose_func),
	  (volume_update_real_volume), (gst_volume_set_volume),
	  (gst_volume_init), (volume_process_double), (volume_process_float),
	  (volume_process_int16), (volume_process_int16_clamp),
	  (volume_set_caps), (volume_transform_ip), (volume_update_volume):
	  * gst/volume/gstvolume.h:
	  Extend float audio support (double) and some int->uint cleanups.

2007-02-20 15:44:32 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
	  (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
	  (sort_end_pads), (gst_decode_group_expose),
	  (gst_decode_group_hide):
	  Don't free groups from the streaming threads. Just put them aside and
	  free them in dispose.

2007-02-20 11:20:52 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (connect_element),
	  (pad_added_group_cb), (gst_decode_group_check_if_blocked),
	  (sort_end_pads), (gst_decode_group_expose):
	  Handle dynamic pads within groups.
	  Sort pads before exposing them in order to make playbin happy.
	  There still is a race with the multiqueue filling up. This should be
	  solved separately.
	  Fixes #398721

2007-02-18 21:02:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
	  Original commit message from CVS:
	  * gst-libs/gst/utils/base-utils.c:
	  * gst-libs/gst/utils/descriptions.c:
	  * gst-libs/gst/utils/install-plugins.c:
	  * gst-libs/gst/utils/missing-plugins.c:
	  Some more docs (and descriptions for two subtitle formats).

2007-02-16 10:19:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/audio.c: Fix documentation.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/audio.c:
	  Fix documentation.

2007-02-16 10:15:46 +0000  Yves Lefebvre <ivanohe@abacom.com>

	  gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
	  Original commit message from CVS:
	  Patch by: Yves Lefebvre  <ivanohe abacom com>
	  * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
	  Don't leak caps. Fixes #408278.

2007-02-15 15:17:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  More docs coverage and some ChangeLog surgery (add missing names)
	  Original commit message from CVS:
	  * ext/cdparanoia/gstcdparanoiasrc.h:
	  * ext/ogg/gstoggdemux.h:
	  * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
	  (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
	  (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
	  * gst-libs/gst/audio/audio.h:
	  * gst-libs/gst/audio/gstaudiofilter.h:
	  * gst-libs/gst/interfaces/videoorientation.h:
	  * gst/adder/gstadder.h:
	  More docs coverage and some ChangeLog surgery (add missing names)

2007-02-15 12:07:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/: Small constifications.
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c:
	  (gst_ximagesink_calculate_pixel_aspect_ratio):
	  * sys/xvimage/xvimagesink.c:
	  (gst_xvimagesink_calculate_pixel_aspect_ratio):
	  Small constifications.

2007-02-15 12:06:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
	  (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
	  (gst_base_audio_sink_async_play),
	  (gst_base_audio_sink_change_state):
	  Answer latency query.
	  Use configured latency when syncing.
	  Fix clock slaving.
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
	  (gst_base_audio_src_query), (gst_base_audio_src_change_state):
	  Fix possible memleak.
	  Implement latency query.
	  Small cleanups.

2007-02-15 11:59:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
	  Ignore errors in reset, these are not fatal. They also grab the element
	  lock which is already taking when this function is called. Fixes
	  #405451.

2007-02-13 13:50:56 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  add header file for easy codec install
	  Original commit message from CVS:
	  add header file for easy codec install

2007-02-13 10:24:13 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
	  Original commit message from CVS:
	  * configure.ac:
	  Remove 'tests/examples/xerror/Makefile' from output files again.

2007-02-13 09:12:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Also crossref against gst-plugins-base-libs.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  Also crossref against gst-plugins-base-libs.

2007-02-12 20:42:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add crossreferences to glib/gobject/gstream docs.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/libs/Makefile.am:
	  * docs/plugins/Makefile.am:
	  Add crossreferences to glib/gobject/gstream docs.
	  * gst-libs/gst/audio/audio.h:
	  Source formatting.
	  * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
	  Add own debug category.

2007-02-12 11:01:04 +0000  René Stadler <mail@renestadler.de>

	  gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
	  Original commit message from CVS:
	  Patch by: René Stadler <mail at renestadler de>
	  * gst-libs/gst/tag/gstvorbistag.c:
	  Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
	  (#403597).

2007-02-12 10:33:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_source):
	  When we have external subtitles and wait for the subtitle decodebin
	  to get up and running, we set up a (sync) bus handler for the
	  subtitle decodebin, so we can stop waiting when it posts an error
	  message. However, we should do that before we set the subtitle
	  decodebin's state to playing, otherwise things are racy and we might
	  miss error messages posted before we had a chance to set up the bus.
	  This should finally fix totem hanging on .txt pseudo-subtitle files.

2007-02-10 19:27:48 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
	  Use gst_gdouble_to_guint64 for conversions.
	  * win32/common/config.h.in:
	  Add a define for GST_INSTALL_PLUGINS_HELPER
	  * win32/common/libgstaudio.def:
	  * win32/common/libgstcdda.def:
	  * win32/common/libgstnetbuffer.def:
	  * win32/common/libgstrtp.def:
	  * win32/common/libgutils.def:
	  Add new exported functions.
	  * win32/vs6/gst_plugins_base.dsw:
	  * win32/vs6/libgstdecodebin.dsp:
	  * win32/vs6/libgstnetbuffer.dsp:
	  * win32/vs6/libgstplaybin.dsp:
	  * win32/vs6/libgstrtp.dsp:
	  * win32/vs6/libgstvorbis.dsp:
	  * win32/vs6/libgstcdda.dsp:
	  * win32/vs6/libgstgdp.dsp:
	  * win32/vs6/libgstutils.dsp:
	  Update and add new project files.

2007-02-10 18:19:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
	  (subrip_remove_unhandled_tags), (parse_subrip):
	  For SubRip (.srt) subtitles, ignore all markup tags we don't
	  handle (like font tags, for example).
	  * tests/check/elements/subparse.c:
	  Add test for this.

2007-02-09 13:28:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery
	  Original commit message from CVS:
	  ChangeLog surgery

2007-02-09 13:16:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (add_fakesink),
	  (gst_decode_bin_change_state):
	  * gst/playback/gstdecodebin2.c: (add_fakesink),
	  (gst_decode_bin_change_state):
	  Don't error out if there is no fakesink in the READY to NULL state
	  change, since when decodebin is re-used, we're only adding the
	  fakesink element in READY to PAUSED.
	  * tests/check/elements/decodebin.c:
	  (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
	  (decodebin_suite):
	  Minimal unit test to make sure we can use the same decodebin
	  instance twice (at least with audiotestsrc input).

2007-02-09 09:58:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
	  Original commit message from CVS:
	  * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
	  Try to get devic-name from device string first, and from handle only
	  as fallback (seems to yield better results and is more robust
	  against buggy probing code on the application side).

2007-02-08 15:43:26 +0000  Julien Puydt <julien.puydt@laposte.net>

	  ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
	  Original commit message from CVS:
	  Based on patch by: Julien Puydt <julien.puydt at laposte net>
	  * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
	  (gst_alsa_find_device_name):
	  * ext/alsa/gstalsa.h:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
	  Improve device-name detection a bit, especially in the case where
	  the device is not actually open (#405020, #405024). Move common code
	  into gstalsa.c instead of duplicating it.

2007-02-07 13:05:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c:
	  Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.

2007-02-06 17:47:32 +0000  Julien Moutte <julien@moutte.net>

	  sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
	  Original commit message from CVS:
	  2007-02-06  Julien MOUTTE  <julien@moutte.net>
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
	  (gst_xvimagesink_get_xv_support),
	  (gst_xvimagesink_xcontext_clear),
	  (gst_xvimagesink_interface_supported),
	  (gst_xvimagesink_probe_get_properties),
	  (gst_xvimagesink_probe_probe_property),
	  (gst_xvimagesink_probe_needs_probe),
	  (gst_xvimagesink_probe_get_values),
	  (gst_xvimagesink_property_probe_interface_init),
	  (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	  (gst_xvimagesink_init), (gst_xvimagesink_class_init),
	  (gst_xvimagesink_get_type):
	  * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
	  for XVAdaptors so that one can choose the adaptor to use with
	  gstreamer-properties.

2007-02-06 14:00:31 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c:
	  Also mention that a conversion from double to float is suboptimal still.

2007-02-06 09:42:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiofilter.c:
	  (gst_audio_filter_class_init), (gst_audio_filter_change_state):
	  Clear our formats structure and free the caps contained in it when
	  shutting down.

2007-02-05 18:39:51 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  gst-libs/gst/audio/gstbaseaudiosink.c
	  Original commit message from CVS:
	  2007-02-05  Andy Wingo  <wingo@pobox.com>
	  * gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_callback): Update basesink->offset so that we
	  pull monotonically increasing offsets instead of, um, seeking back
	  to 0 each time. Fixes alsasrc ! alsasink!

2007-02-05 11:44:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
	  Original commit message from CVS:
	  * gst/videoscale/gstvideoscale.c:
	  A width and height of 1 makes us crash, so increase minimum size to
	  2x2 pixels until someone feels like fixing this (#404512).

2007-02-04 16:23:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
	  Original commit message from CVS:
	  * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
	  Add small test to make sure request pads are cleaned up properly
	  even if oggmux never changes state out of NULL.

2007-02-04 14:11:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
	  Original commit message from CVS:
	  * tests/check/libs/utils.c: (GST_START_TEST):
	  Fix unit test. Turns out things work much better when you
	  NULL-terminate string arrays. Should make p5 build bot happy again.

2007-02-03 23:28:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/Makefile.am:
	  * gst-libs/gst/audio/gstaudiofiltertemplate.c:
	  (gst_audio_filter_template_base_init),
	  (gst_audio_filter_template_class_init),
	  (gst_audio_filter_template_init),
	  (gst_audio_filter_template_set_property),
	  (gst_audio_filter_template_get_property),
	  (gst_audio_filter_template_setup),
	  (gst_audio_filter_template_filter),
	  (gst_audio_filter_template_filter_inplace), (plugin_init):
	  Oops, forgot to commit fixed-up example.

2007-02-03 20:19:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
	  (gst_audio_filter_class_init), (gst_audio_filter_init),
	  (gst_audio_filter_set_caps),
	  (gst_audio_filter_class_add_pad_templates):
	  * gst-libs/gst/audio/gstaudiofilter.h:
	  Port GstAudioFilter to 0.10. This change technically breaks
	  API and ABI (and thus also every library developer's heart),
	  but seems justifiable on the grounds that the base class was
	  completely unusable before (ie. would crash immediately when
	  actually used). Fixes #403963 (and eventually also #403572).
	  Also document all of this a bit.

2007-02-03 14:26:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
	  Original commit message from CVS:
	  * gst-libs/gst/utils/install-plugins.c:
	  (gst_install_plugins_spawn_child):
	  * tests/check/libs/utils.c:
	  (test_base_utils_install_plugins_do_callout):
	  Lowering log level to see why things fail on the p5 build bot;
	  fix some typos in unit test messages.

2007-02-03 13:59:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
	  Original commit message from CVS:
	  * tests/check/libs/utils.c:
	  (test_base_utils_install_plugins_do_callout):
	  Don't hard-code temp directory for test helper; use GLib functions
	  to write out file and do error checking etc.

2007-02-02 20:42:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
	  Original commit message from CVS:
	  * gst-libs/gst/utils/Makefile.am:
	  * gst-libs/gst/utils/base-utils.h:
	  * gst-libs/gst/utils/install-plugins.c:
	  (gst_install_plugins_context_set_xid),
	  (gst_install_plugins_context_new),
	  (gst_install_plugins_context_free),
	  (gst_install_plugins_get_helper),
	  (gst_install_plugins_spawn_child),
	  (gst_install_plugins_return_from_status),
	  (gst_install_plugins_installer_exited),
	  (gst_install_plugins_async), (gst_install_plugins_sync),
	  (gst_install_plugins_return_get_name),
	  (gst_install_plugins_installation_in_progress):
	  * gst-libs/gst/utils/install-plugins.h:
	  API: add API for applications to initiate installation of missing
	  plugins, ie. gst_install_plugins_async() primarily.
	  Based on libgimme-codec by Ryan Lortie.
	  * configure.ac:
	  Add --with-install-plugins-helper configure option so distros can specify
	  the path of the helper script or program to call when plugin installation
	  is requested (distros: please do any argument munging in this helper
	  script instead of patching GStreamer to pass arguments differently
	  to another program directly).
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Build and document new API.
	  * tests/check/libs/utils.c: (result_cb),
	  (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
	  (libgstbaseutils_suite):
	  Some simple checks for the new API.

2007-02-02 14:44:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (test_float_conversion):
	  Add small test for 32bit float <=> 64bit float conversion (works
	  only one way so far, 32=>64 produces structured noise).

2007-02-02 11:21:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c:
	  (set_structure_widths_32_and_64), (make_lossless_changes):
	  We don't support floats with a width of 40, 48 or 56 bits.

2007-02-02 09:48:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c: (float), (double),
	  (audio_convert_get_func_index):
	  * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
	  (make_lossless_changes):
	  Support for 64-bit float audio in audioconvert (#339837)

2007-02-01 18:50:08 +0000  Holger Wansing <linux@wansing-online.de>

	  po/: Add German translation (#352069).
	  Original commit message from CVS:
	  Patch by: Holger Wansing  <linux wansing-online de>
	  * po/LINGUAS:
	  * po/de.po:
	  Add German translation (#352069).

2007-02-01 17:52:39 +0000  Sebastian Dröge <slomo@circular-chaos.org>

	  ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
	  Original commit message from CVS:
	  reviewed by: Wim Taymans <wim@fluendo.com>
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
	  (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
	  Use newly added GstCollectPads API to free the allocated resources in
	  the GstOggPad structures (#402393).

2007-01-31 15:58:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gen_vis_element):
	  Add audioresample+audioconvert in front of the visualisation
	  element, so that elements like libvisual 0.4 that don't support all
	  samplerates can work.
	  Fixes: #402505

2007-01-30 19:19:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
	  (gst_play_base_bin_get_streaminfo_value_array):
	  Take some locks and make a copy of the streaminfo value array we
	  maintain while holding the lock, so that the application can
	  retrieve the stream-info as a value array in a thread-safe way.

2007-01-30 11:29:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c:
	  Don't fail on 0 sized buffers. Fixes #396835.

2007-01-29 21:13:07 +0000  David Schleef <ds@schleef.org>

	  gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c:
	  Detect BBCD as video/x-dirac, so we can play raw dirac
	  streams.

2007-01-29 18:14:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
	  Original commit message from CVS:
	  * ext/theora/theoraenc.c: (theora_enc_chain):
	  Check return value of theora_encode_header(), or we might try to
	  allocate a random number of bytes. theora_encode_header() can fail
	  if libtheora has been compiled with encoding support disabled.
	  Fixes #398110.

2007-01-29 10:53:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/gst/.cvsignore: Do as buildbot says.
	  Original commit message from CVS:
	  * tests/check/gst/.cvsignore:
	  Do as buildbot says.

2007-01-29 10:25:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_src_setcaps):
	  Fix strides in libvisual. Gst uses X strides.
	  Inspired by: <ed at catmur dot co dot uk> and
	  <tim at centricular dot net>
	  Fixes #401118.

2007-01-27 13:32:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
	  (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
	  (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
	  (gst_ogg_demux_perform_seek),
	  (gst_ogg_demux_bisect_forward_serialno),
	  (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
	  (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
	  (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
	  (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
	  * ext/ogg/gstoggdemux.h:
	  Properly propagate streaming errors when we are scanning the file for
	  chains so that we don't crash when shut down. Might fix some crashers
	  when quickly switching oggs in RB such as #332503 and #378436.

2007-01-26 12:44:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
	  Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
	  error code as well.

2007-01-25 16:02:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (remove_source):
	  Don't try to disconnect a signal from a finalized object.

2007-01-25 14:29:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
	  Cast lock macro parameters to make sure we're actually accessing the
	  lock member at the right class level. Free list itself in _dispose()
	  as well and NULL it in case dispose gets called multiple times.

2007-01-25 14:02:37 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c:
	  (gst_decode_bin_dispose),(gst_decode_bin_finalize):
	  Free GstDecodeGroups no longer used.
	  (gst_decode_group_expose):
	  Don't unlock too many times !
	  (deactivate_free_recursive):
	  Free iterator once we're done with it.
	  Fix for recursively deactivating elements (stop at ghostpads).

2007-01-25 12:24:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (handoff):
	  Fix up caps on the frame buffer before we save it and potentially
	  make it accessible to other threads via g_object_get; also use
	  gst_buffer_replace() instead of gst_mini_object_replace().

2007-01-25 12:06:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
	  Make getting the current frame thread-safe.

2007-01-25 11:48:10 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
	  (gst_decode_group_new), (gst_decode_group_free):
	  Set queues to bigger sizes to cope with HD contents.
	  Fix some mutex freeing and add comment about MT safe methods.

2007-01-24 12:51:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
	  (gst_text_overlay_text_event):
	  Don't unnecessarily ref (and then leak) upstream events if the text
	  pad is not linked. Fixes #399948.
	  * tests/check/gst-plugins-base.supp:
	  Add suppression for pango on edgy/x86 for textoverlay test.

2007-01-24 12:10:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Add some more fixed payloads.

2007-01-23 18:39:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
	  Error out properly if we get an error from libogg while reading the
	  BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).

2007-01-23 17:49:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstdecodebin2.c: Don't leak mutex.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
	  Don't leak mutex.
	  * tests/check/elements/playbin.c:
	  (test_sink_usage_video_only_stream),
	  (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
	  (test_suburi_error_wrongproto), (test_missing_urisource_handler),
	  (test_missing_suburisource_handler),
	  (test_missing_primary_decoder), (playbin_suite):
	  Run all tests once with decodebin and once with decodebin2.
	  One test does not pass yet with decodebin2.

2007-01-23 14:30:28 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
	  Fix the cases where oggmux doesn't properly figure out that all
	  sinkpads have gone EOS, and therefore doesn't push out the remaining
	  buffers and the final EOS event.
	  Fixes #363379

2007-01-23 13:19:19 +0000  Julien Moutte <julien@moutte.net>

	  sys/: Don't lock on navigation event push, just on keysym to string.
	  Original commit message from CVS:
	  2007-01-23  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	  Don't lock on navigation event push, just on keysym to string.
	  Fixes #397673 again.

2007-01-22 17:37:38 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstdecodebin2.c: Cleanups.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
	  (get_current_group), (group_demuxer_event_probe),
	  (gst_decode_group_expose), (deactivate_free_recursive),
	  (gst_decode_group_free):
	  Cleanups.
	  Don't forget to emit 'no-more-pads' once a group is exposed.
	  Cleanup elements from a DecodeGroup once we remove it.
	  Protect call to gst_decode_group_expose() with the decodebin lock.

2007-01-22 13:16:42 +0000  Julien Moutte <julien@moutte.net>

	  sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
	  Original commit message from CVS:
	  2007-01-22  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	  Looking at Xorg code i can't figure out if that XKeysymToString
	  function is thread sensible or not. Lock it just in case as
	  recommended by Radek Doulik <rodo at ximian dot com>.

2007-01-22 13:10:13 +0000  Julien Moutte <julien@moutte.net>

	  sys/: Lock that X Call as well. Fixes #397673.
	  Original commit message from CVS:
	  2007-01-22  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
	  Lock that X Call as well. Fixes #397673.

2007-01-22 12:03:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
	  Don't go into an endless loop if the file starts with 00 00 01 2X,
	  like quicktime redirect files might. Fixes #396042.
	  * tests/check/Makefile.am:
	  * tests/check/gst/.cvsignore:
	  * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
	  (typefindfunctions_suite):
	  Add unit test for the above.

2007-01-22 10:27:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  On second thought, use "depth" field rather than "bpp" field.

2007-01-22 09:23:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Camtasia caps apparently need a bpp field (#398875).

2007-01-19 19:09:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_subtitle),
	  (gen_source_element), (gst_play_base_bin_change_state):
	  Attempt at a better error message in case we don't have the required
	  URI handler installed; post missing-plugin message also when we're
	  missing an URI handler for the subtitle URI; clean up properly also
	  when an error occurs and we never made it to PAUSED state.
	  * tests/check/elements/playbin.c: (GST_START_TEST),
	  (playbin_suite):
	  Check that we're also getting a missing-plugin messsage for a
	  missing subtitle URI handler (and clean up properly).

2007-01-19 18:47:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Plug a few reference leaks.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
	  Plug a few reference leaks.

2007-01-19 12:23:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  Lower probability a bit if the marker isn't right at the start,
	  to decrease the chance of false positives.

2007-01-19 11:31:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  Small mpeg2 system stream typefinding improvement: make typefinder
	  probe a bit into the stream instead of just looking for a marker
	  at the beginning. Fixes #397810.

2007-01-18 16:23:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
	  Original commit message from CVS:
	  * gst/audioconvert/gstchannelmix.c:
	  Remove compatibility cruft for prehistoric GLib versions.

2007-01-17 16:11:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
	  Original commit message from CVS:
	  * gst/playback/Makefile.am:
	  * gst/playback/gstdecodebin.c: (close_pad_link):
	  * gst/playback/gstdecodebin2.c: (analyze_new_pad):
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	  (gst_play_base_bin_handle_message_func), (unknown_type):
	  Let decodebin be the element to post missing-plugin messages for
	  missing decoders (rather than playbin); make playbin implement
	  GstBin::handle_message so we can suppress missing-plugin messages
	  for types we're not handling on purpose (don't want to bring up an
	  installer in those cases).

2007-01-16 19:37:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Fix potentially unaligned access (#397207).
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  * gst-libs/gst/tag/gstvorbistag.c:
	  (gst_tag_list_to_vorbiscomment_buffer):
	  * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
	  Fix potentially unaligned access (#397207).

2007-01-16 12:17:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (set_scale), (update_scale),
	  (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
	  (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
	  (main):
	  Allow to toggle looping while it plays. Fix callback prototype. Clean
	  up code a bit more. Add copyright header.

2007-01-16 11:41:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
	  Red and blue mask was swapped (spotted by Dan Williams).

2007-01-15 13:58:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/tag/: Use new beats-per-minute tag from core.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  Use new beats-per-minute tag from core.

2007-01-15 11:30:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  Add new files with translatable strings, so they actually make it
	  into the template file one day.

2007-01-12 21:19:35 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	  gst-libs/gst/audio/gstbaseaudiosink.c
	  Original commit message from CVS:
	  2007-01-12  Andy Wingo  <wingo@pobox.com>
	  * gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
	  (gst_base_audio_sink_activate_pull): Remove the handwavey nego
	  stuff, as the base class handles this now. Actually tell the ring
	  buffer to start.
	  (gst_base_audio_sink_callback): Cast the ring buffer correctly.
	  How did this work before? Maybe I'm not as awesome a programmer as
	  I think.
	  * gst-libs/gst/audio/gstbaseaudiosrc.c
	  (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
	  of a pad function.

2007-01-12 18:08:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
	  Original commit message from CVS:
	  * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
	  Remove more fields so that the application can better blacklist
	  formats that have been tried before.

2007-01-12 17:43:40 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  add latest files
	  Original commit message from CVS:
	  add latest files

2007-01-12 12:47:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/mixerutils.h:
	  Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
	  used when compiling with c++ compilers as well.

2007-01-12 09:45:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Fix comment.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c:
	  Fix comment.

2007-01-11 13:12:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (post_missing_element_message),
	  (gen_video_element), (gen_text_element), (gen_audio_element),
	  (gen_vis_element):
	  Post missing-plugin messages also when we error out because
	  converters, textoverlay or auto*sinks are missing (#161922).

2007-01-10 16:08:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
	  (is_demuxer_element), (new_caps):
	  * gst/playback/gstplaybasebin.c: (source_new_pad):
	  Fix the case where we try to ref a NULL element when we delay a link
	  because of unfixed caps.
	  Set the state of autoplugged decodebins to PAUSED.
	  RTSP now works in playbin, we can remove it from the blacklist.

2007-01-09 14:33:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
	  Original commit message from CVS:
	  * gst/playback/Makefile.am:
	  * gst/playback/gstplaybasebin.c: (string_arr_has_str),
	  (unknown_type), (setup_subtitle), (gen_source_element):
	  * gst/playback/gstplaybin.c: (plugin_init):
	  Post missing-plugin messages on the bus for missing sources and
	  missing decoders/demuxers/depayloaders; fix error code used when
	  we're missing an URI handler source; for media types that we are not
	  handling on purpose at the moment, don't print "don't know how to
	  handle xyz" messages to the terminal or post missing-plugin
	  messages on the bus.
	  * tests/check/elements/playbin.c: (create_playbin),
	  (GST_START_TEST), (gst_codec_src_uri_get_type),
	  (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
	  (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
	  (gst_codec_src_init_type), (gst_codec_src_base_init),
	  (gst_codec_src_create), (gst_codec_src_class_init),
	  (gst_codec_src_init), (plugin_init), (playbin_suite):
	  Add some tests for the missing-plugin stuff.

2007-01-09 14:20:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  API: add new libgstbaseutils library with functions
	  Original commit message from CVS:
	  * configure.ac:
	  * gst-libs/gst/Makefile.am:
	  * gst-libs/gst/utils/Makefile.am:
	  * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
	  * gst-libs/gst/utils/base-utils.h:
	  * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
	  (find_format_info), (caps_are_rtp_caps),
	  (gst_base_utils_get_source_description),
	  (gst_base_utils_get_sink_description),
	  (gst_base_utils_get_decoder_description),
	  (gst_base_utils_get_encoder_description),
	  (gst_base_utils_get_element_description),
	  (gst_base_utils_add_codec_description_to_tag_list),
	  (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
	  * gst-libs/gst/utils/descriptions.h:
	  * gst-libs/gst/utils/missing-plugins.c:
	  (missing_structure_get_type), (copy_and_clean_caps),
	  (gst_missing_uri_source_message_new),
	  (gst_missing_uri_sink_message_new),
	  (gst_missing_element_message_new),
	  (gst_missing_decoder_message_new),
	  (gst_missing_encoder_message_new),
	  (missing_structure_get_string_detail),
	  (missing_structure_get_caps_detail),
	  (gst_missing_plugin_message_get_installer_detail),
	  (gst_missing_plugin_message_get_description),
	  (gst_is_missing_plugin_message):
	  * gst-libs/gst/utils/missing-plugins.h:
	  API: add new libgstbaseutils library with functions
	  - to create and parse missing-plugins messages
	  - that provide (translated) descriptions for caps/decoders/sources/etc.
	  Closes #392393.
	  * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	  * pkgconfig/gstreamer-plugins-base.pc.in:
	  Add new lib.
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Generate docs for new lib and API.
	  * tests/check/Makefile.am:
	  * tests/check/libs/.cvsignore:
	  * tests/check/libs/utils.c: (missing_msg_check_getters),
	  (GST_START_TEST), (libgstbaseutils_suite):
	  Add some basic unit tests.

2007-01-09 13:35:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
	  Original commit message from CVS:
	  * ext/ogg/Makefile.am:
	  Dist gstoggdemux.h to fix 'make distcheck'.
	  * sys/v4l/Makefile.am:
	  Fix 'make distcheck' even more.

2007-01-09 12:30:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Added docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
	  (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
	  (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
	  (gst_ogg_demux_perform_seek):
	  * ext/ogg/gstoggdemux.h:
	  Added docs.
	  Add some more comments.
	  Small cleanups.

2007-01-09 11:15:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Small documentation updates/fixes
	  Original commit message from CVS:
	  * ext/theora/theoradec.c:
	  * ext/vorbis/vorbisdec.c:
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_commit_full):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  * gst-libs/gst/rtp/gstrtpbuffer.c:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  Small documentation updates/fixes

2007-01-09 10:37:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
	  Original commit message from CVS:
	  * configure.ac:
	  Require core CVS HEAD for Andy's basesrc/sink API additions.

2007-01-08 14:01:23 +0000  Günter Thelen <daedalus.inc@gmx.net>

	  gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
	  Original commit message from CVS:
	  Patch by: Günter Thelen  <daedalus dot inc at gmx net>
	  * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
	  (plugin_init):
	  Add typefinder for flac-in-ogg in conformance with the ogg-mapping
	  on flac.sf.net (there appear to be other versions of the first
	  ogg page in the wild) (#391365).

2007-01-08 13:32:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Check if localtime_r() is available.
	  Original commit message from CVS:
	  * configure.ac:
	  Check if localtime_r() is available.
	  * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
	  If localtime_r() is not available, fall back to localtime(). Should
	  fix build on MingW (#393310).

2007-01-08 12:30:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (parse_mdvdsub):
	  * gst/subparse/gstsubparse.h:
	  Remove spurious 1000 subtrahend when calculating the timestamp from
	  the frame number and the frame rate . Also, use the frames/second
	  value specified in the first line of the file, if one is specified
	  there. Should fix #357503.
	  * tests/check/elements/subparse.c: (do_test),
	  (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
	  (subparse_suite):
	  Add some basic unit tests for the microdvd subtitle format.

2007-01-07 21:53:38 +0000  Young-Ho Cha <ganadist@chollian.net>

	  sys/xvimage/xvimagesink.c: Fixes : #390076.
	  Original commit message from CVS:
	  2007-01-07  Julien MOUTTE  <julien@moutte.net>
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
	  (gst_xvimagesink_xvimage_put),
	  (gst_lookup_xv_port_from_adaptor),
	  (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
	  (gst_xvimagesink_set_xwindow_id),
	  (gst_xvimagesink_set_event_handling),
	  (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
	  (gst_xvimagesink_init), (gst_xvimagesink_class_init):
	  Patch by : Young-Ho Cha <ganadist at chollian dot net>
	  Fixes : #390076.
	  Add an adaptor property to select a specific XV adaptor.
	  * sys/xvimage/xvimagesink.h:

2007-01-07 18:50:13 +0000  Julien Moutte <julien@moutte.net>

	  sys/: Use flow_lock much more to protect every access to xwindow.
	  Original commit message from CVS:
	  2007-01-07  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	  (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
	  (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
	  (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
	  (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
	  (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
	  (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
	  (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
	  (gst_xvimagesink_change_state),
	  (gst_xvimagesink_set_xwindow_id),
	  (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
	  Use flow_lock much more to protect every access to xwindow.
	  Try to catch erros while creating images in case some drivers
	  are
	  just generating an XError when the requested image is too big.
	  Should fix : #354698, #384008, #384060.
	  * tests/icles/stress-xoverlay.c: (cycle_window),
	  (create_window):
	  Implement some stress testing of setting window xid.

2007-01-07 10:33:55 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/common/libgsaudio.def: Add new exported function.
	  Original commit message from CVS:
	  * win32/common/libgsaudio.def:
	  Add new exported function.
	  * win32/common/libgstogg.dsp:
	  Add gstoggaviparse.c to the build.
	  * win32/common/libgstvideoscale.dsp:
	  Add vs_4tap.c to the build.
	  * win32/common/libgstvorbis.dsp:
	  Add vorbistag.c to the build.

2007-01-06 17:28:40 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	  gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
	  Original commit message from CVS:
	  2007-01-06  Andy Wingo  <wingo@pobox.com>
	  * gst-libs/gst/audio/gstbaseaudiosink.c
	  (gst_base_audio_sink_class_init)
	  (gst_base_audio_sink_init):
	  (gst_base_audio_sink_activate_pull): Add an activate_pull function
	  to baseaudiosink, and tell basesink that we can work in pull mode.
	  This way the ring buffer thread drives the pipeline directly, if
	  pull mode is possible. There is some lingering nastiness regarding
	  capsnego, however.
	  (gst_base_audio_sink_callback): Implement the callback to pull
	  data. This interface is a bit light, though -- it should get a
	  GstFlowReturn return value at least.

2007-01-05 19:43:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Printf format and missing argument fixes.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  * gst/playback/gstdecodebin2.c:
	  (gst_decode_group_check_if_blocked):
	  Printf format and missing argument fixes.

2007-01-05 18:57:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
	  Original commit message from CVS:
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
	  (gst_ogm_parse_change_state):
	  Activate pads before adding them to the element.

2007-01-05 16:02:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
	  Original commit message from CVS:
	  * tests/examples/seek/scrubby.c: (main):
	  * tests/examples/seek/seek.c: (main):
	  Call g_thread_init() first thing in main() (see #391278).

2007-01-05 12:19:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/libs/.cvsignore:
	  * tests/check/libs/netbuffer.c: (GST_START_TEST),
	  (netbuffer_suite):
	  Add test for GstNetBuffer + gst_buffer_copy(). Disabled
	  for the time being, since it's broken, see #393099.

2007-01-05 12:13:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Update to use GST_PLUGINS_BASE_CFLAGS as well.

2007-01-04 12:49:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
	  Original commit message from CVS:
	  * configure.ac:
	  split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
	  so that GST_BASE_CFLAGS can go inbetween them, making sure
	  we use uninstalled gst-libs headers
	  * docs/libs/Makefile.am:
	  * ext/alsa/Makefile.am:
	  * ext/cdparanoia/Makefile.am:
	  * ext/gnomevfs/Makefile.am:
	  * ext/libvisual/Makefile.am:
	  * ext/ogg/Makefile.am:
	  * ext/theora/Makefile.am:
	  * ext/vorbis/Makefile.am:
	  * gst-libs/gst/audio/Makefile.am:
	  * gst-libs/gst/cdda/Makefile.am:
	  * gst-libs/gst/interfaces/Makefile.am:
	  * gst-libs/gst/riff/Makefile.am:
	  * gst-libs/gst/rtp/Makefile.am:
	  * gst-libs/gst/tag/Makefile.am:
	  * gst/adder/Makefile.am:
	  * gst/audioconvert/Makefile.am:
	  * gst/audiorate/Makefile.am:
	  * gst/audioresample/Makefile.am:
	  * gst/playback/Makefile.am:
	  * gst/tcp/Makefile.am:
	  * gst/videoscale/Makefile.am:
	  * gst/volume/Makefile.am:
	  * sys/ximage/Makefile.am:
	  * sys/xvimage/Makefile.am:
	  * tests/icles/Makefile.am:
	  adapt

2007-01-04 11:30:53 +0000  Julien Moutte <julien@moutte.net>

	  Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
	  Original commit message from CVS:
	  2007-01-04  Julien MOUTTE  <julien@moutte.net>
	  * gst-libs/gst/interfaces/xoverlay.c:
	  (gst_x_overlay_handle_events):
	  * gst-libs/gst/interfaces/xoverlay.h:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
	  (gst_ximagesink_set_xwindow_id),
	  (gst_ximagesink_set_event_handling),
	  (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
	  (gst_ximagesink_get_property), (gst_ximagesink_init),
	  (gst_ximagesink_class_init):
	  * sys/ximage/ximagesink.h:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
	  (gst_xvimagesink_set_xwindow_id),
	  (gst_xvimagesink_set_event_handling),
	  (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
	  (gst_xvimagesink_get_property), (gst_xvimagesink_init),
	  (gst_xvimagesink_class_init):
	  * sys/xvimage/xvimagesink.h:
	  * tests/icles/stress-xoverlay.c: (toggle_events),
	  (create_window):
	  Add a method to the XOverlay interface to allow disabling of
	  event handling in x[v]imagesink elements. This will let X events
	  propagate to parent windows which can be usefull in some cases.
	  Be carefull that the application is then responsible of pushing
	  navigation events and expose events to the video sink.
	  Fixes: #387138.

2007-01-03 15:45:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  * tests/check/libs/tag.c: (GST_START_TEST):
	  Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
	  (fixes #392070).

2007-01-01 18:59:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Dist design docs.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/Makefile.am:
	  * docs/design/Makefile.am:
	  Dist design docs.

2006-12-27 17:15:35 +0000  Julien Moutte <julien@moutte.net>

	  docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
	  Original commit message from CVS:
	  2006-12-27  Julien MOUTTE  <julien@moutte.net>
	  * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
	  documentation
	  typo. Fixes: #390063.

2006-12-27 12:08:13 +0000  Julien Moutte <julien@moutte.net>

	  sys/: Plug a caps leak.
	  Original commit message from CVS:
	  2006-12-27  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
	  caps leak.
	  * win32/common/config.h: Updated.

2006-12-22 12:10:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
	  Original commit message from CVS:
	  * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
	  (setup_gdpdepay_streamheader):
	  * tests/check/elements/gdppay.c: (cleanup_gdppay),
	  (setup_gdppay_streamheader):
	  Fix the dp tests, but activating the pads for the streamheader tests
	  too and cleaning up conditionaly

2006-12-22 11:09:34 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/avcodec.h:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
	  (gst_ffmpegcsp_avpicture_fill):
	  * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
	  (img_get_alpha_info):
	  Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
	  other end of the word. Fixes: #387073.
	  Add some inconsequential branch hints in a couple of places.

2006-12-21 12:30:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_caps_to_smpfmt):
	  The "signed" field in raw audio caps is of boolean type, trying to
	  extract the value with _get_int() will fail (fix to keep in sync with
	  the copy in gst-ffmpeg)

2006-12-21 08:12:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/: consistent pad (de)activation
	  Original commit message from CVS:
	  * tests/check/elements/audioresample.c: (cleanup_audioresample):
	  * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
	  * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
	  (cleanup_gdpdepay):
	  * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
	  * tests/check/elements/subparse.c: (teardown_subparse):
	  * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
	  * tests/check/elements/videorate.c: (cleanup_videorate):
	  * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
	  * tests/check/elements/volume.c: (cleanup_volume):
	  * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
	  (cleanup_vorbisdec):
	  * tests/check/elements/vorbistag.c: (setup_vorbistag),
	  (cleanup_vorbistag):
	  consistent pad (de)activation

2006-12-20 10:29:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Forgot to register the extensions.

2006-12-20 09:25:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
	  (plugin_init):
	  Add typefinder for VIVO files (my christmas present to the 90s).

2006-12-16 13:59:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (type_found):
	  Special-case the text/plain media type: we only want to recognise it
	  as a 'raw' decoded media type if it comes from a demuxer or subtitle
	  parser, but not if the entire stream is of text/plain type. If the
	  entire stream is text/plain, we should just error out.
	  This fixes playback of audio files with lyrics in totem. Totem can't
	  distinguish between text files and subtitle files and passes any
	  .txt file with the same basename as the main file to playbin as
	  suburi, and playbin will then throw a 'subtitle found, but no video
	  stream' error, which isn't entirely helpful. See #380342.
	  Also, with this change we'll show a slightly more correct error
	  message in case totem passes a playlist file to us (although a
	  custom error message wording instead of the default text would
	  probably not be a bad idea either).
	  Same problem also needs to be fixed for playbin+decodebin2.
	  * tests/check/Makefile.am:
	  * tests/check/elements/decodebin.c: (src_handoff_cb),
	  (decodebin_new_decoded_pad_cb), (GST_START_TEST),
	  (decodebin_suite):
	  Add simple unit test for decodebin for the above.

2006-12-16 12:22:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
	  Refuse to change state to READY when we failed to create any of the
	  required elements in our instance init function.

2006-12-15 10:52:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Small docs fixes/updates.
	  * gst-libs/gst/video/gstvideosink.h:
	  Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
	  from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
	  removed from the base sink API between 0.9.6 and 0.9.7).
	  API: add GST_VIDEO_SINK_CAST and use it for the height/width
	  accessor macros, so we don't do a runtime GObject type check every
	  time we use them.

2006-12-15 00:20:37 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add doap file
	  Original commit message from CVS:
	  * Makefile.am:
	  * gst-plugins-base.doap:
	  * gst-plugins-base.spec.in:
	  add doap file

2006-12-09 15:12:38 +0000  Jens Granseuer <jensgr@gmx.net>

	  Declare variables at the beginning of a block. Fixes #383195.
	  Original commit message from CVS:
	  Patch by: Jens Granseuer <jensgr at gmx net>
	  * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_handle_frame_based_buffer),
	  (gst_base_rtp_audio_payload_handle_sample_based_buffer):
	  * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
	  Declare variables at the beginning of a block. Fixes #383195.

2006-12-07 02:38:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Bump version nano - back to CVS.
	  Original commit message from CVS:
	  * configure.ac:
	  Bump version nano - back to CVS.

=== release 0.10.11 ===

2006-12-07 02:30:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: releasing 0.10.11, "Dumb things"
	  Original commit message from CVS:
	  === release 0.10.11 ===
	  2006-12-06  Jan Schmidt <thaytan@mad.scientist.com>
	  * configure.ac:
	  releasing 0.10.11, "Dumb things"

2006-12-05 12:44:00 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
	  (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
	  Handle the case where an element has multiple pads with
	  unfixed caps as well as still possibly producing more dynamic
	  pads by storing each case as a distinct entry in the dynamic list.
	  Fixes #38223 again.

2006-12-04 13:02:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (close_pad_link):
	  Fix #382223, add more dynamic caps handling.

2006-12-01 11:35:57 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	* po/.gitignore:
	  Ignore all pot files
	  Original commit message from CVS:
	  Ignore all pot files

2006-12-01 10:36:50 +0000  Michael Smith <msmith@xiph.org>

	  gst/audiorate/gstaudiorate.c: Delete bad debug code.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  Delete bad debug code.
	  Fixes #381219

2006-12-01 10:27:54 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  Fix compilation on win32 under VS8
	  Original commit message from CVS:
	  * gst/videoscale/vs_4tap.c:
	  * win32/MANIFEST:
	  * win32/common/config.h:
	  * win32/vs8/libgstvideoscale.vcproj:
	  Fix compilation on win32 under VS8
	  Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
	  Partially fixes #381175

2006-11-30 23:46:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-11-30 12:50:42 +0000  Michael Smith <msmith@xiph.org>

	  tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
	  Original commit message from CVS:
	  * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
	  (GST_START_TEST):
	  It would be very bad if, after a discont buffer, we thought every
	  single following buffer was also discont. So, add to the test to
	  ensure that this isn't the case.
	  * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
	  ... it was the case. So fix it.

2006-11-28 16:43:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Improve debug.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (check_queue_event):
	  Improve debug.
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
	  Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
	  padtemplate caps. Refixes #357577.

2006-11-28 16:21:27 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (check_queue_event),
	  (queue_threshold_reached), (queue_out_of_data),
	  (gen_preroll_element):
	  Add event probe to see when EOS is in a queue and we can disable the
	  underrun signals. Fixes #357577.

2006-11-28 14:40:39 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/: New decodebin2 element.
	  Original commit message from CVS:
	  * gst/playback/Makefile.am:
	  * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
	  (_gst_boolean_accumulator), (gst_decode_bin_class_init),
	  (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
	  (gst_decode_bin_init), (gst_decode_bin_dispose),
	  (gst_decode_bin_finalize), (gst_decode_bin_set_property),
	  (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
	  (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
	  (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
	  (connect_element), (expose_pad), (type_found),
	  (pad_added_group_cb), (pad_removed_group_cb),
	  (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
	  (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
	  (are_raw_caps), (multi_queue_overrun_cb),
	  (multi_queue_underrun_cb), (gst_decode_group_new),
	  (get_current_group), (group_demuxer_event_probe),
	  (gst_decode_group_control_demuxer_pad),
	  (gst_decode_group_control_source_pad),
	  (gst_decode_group_check_if_blocked),
	  (gst_decode_group_check_if_drained), (gst_decode_group_expose),
	  (gst_decode_group_hide), (gst_decode_group_free),
	  (gst_decode_group_set_complete), (source_pad_blocked_cb),
	  (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
	  (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
	  (plugin_init):
	  New decodebin2 element.
	  Closes #370092
	  * gst/playback/gstplay-marshal.list:
	  Added marshallers for new signals in decodebin2
	  * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
	  Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
	  is set.

2006-11-28 10:45:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_source),
	  (gst_play_base_bin_change_state):
	  Disable rtsp:// uris for the release, it's not good enough yet.
	  Remove unused var.

2006-11-26 16:39:41 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/theoradec.c: Implement reverse playback.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (gst_theora_dec_reset),
	  (theora_dec_push_forward), (theora_dec_push_reverse),
	  (theora_handle_data_packet), (theora_dec_decode_buffer),
	  (theora_dec_flush_decode), (theora_dec_chain_reverse),
	  (theora_dec_chain_forward), (theora_dec_chain):
	  Implement reverse playback.
	  * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
	  (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
	  (vorbis_dec_chain_forward):
	  Clear buffers used for reverse playback in _reset.
	  No need to set the eos flag, we clip samples using the segment.

2006-11-24 15:40:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Some cleanups.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
	  (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
	  (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
	  (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
	  Some cleanups.
	  Handle continued pages in reverse mode.

2006-11-24 15:39:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.c: Small cleanups.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
	  (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
	  (vorbis_dec_flush_decode):
	  Small cleanups.
	  Don't try to add invalid timestamps.
	  Clipping will unref the buffer.

2006-11-24 08:56:10 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: remove obsolete _factory_init protos
	  Original commit message from CVS:
	  * gst/adder/gstadder.h:
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  remove obsolete _factory_init protos

2006-11-24 08:35:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/xvimage/xvimagesink.c: Fix spacing in debug message.
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
	  Fix spacing in debug message.

2006-11-23 11:07:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
	  (gst_ogg_demux_chain):
	  Don't just ignore return values from _pad_push().
	  Small debug improvements.

2006-11-23 11:02:11 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
	  If our incoming buffer is marked as DISCONT, then increment the page
	  number (so that the discontinuity is marked in the final ogg
	  bitstream) and flush the previous page.

2006-11-22 14:34:03 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
	  Original commit message from CVS:
	  * ext/theora/gsttheoraenc.h:
	  * ext/theora/theoraenc.c: (gst_theora_enc_init),
	  (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
	  (theora_buffer_from_packet), (theora_enc_is_discontinuous),
	  (theora_enc_chain), (theora_enc_change_state):
	  Mark discontinuities of > 3/4 of a frame, reinit encoder.
	  * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
	  (GST_START_TEST), (theoraenc_suite):
	  Enable discontinuity test, fix it.

2006-11-21 18:39:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
	  (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
	  (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
	  (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
	  (gst_text_overlay_change_state):
	  * ext/pango/gsttextoverlay.h:
	  Some textoverlay fixes: for one, in the video chain function,
	  actually wait for a text buffer to come in if there is none at the
	  moment and there should be one; also, deal more gracefully with
	  incoming buffers that do not have a timestamp or duration; discard
	  text buffer when not needed any longer. Fixes #341681.
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/textoverlay.c:
	  (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
	  (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
	  (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
	  (test_video_waits_for_text_send_text_newsegment_thread),
	  (test_video_waits_for_text_shutdown_element),
	  (test_render_continuity_push_video_buffers_thread),
	  (textoverlay_suite):
	  Add some unit tests for textoverlay.

2006-11-21 09:29:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
	  Avoid integer underflow when the found probability for mp3 is
	  smaller than the 'penalty' we subtract if there's not a clean
	  mp3 header sync at offset 0.

2006-11-21 08:17:16 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  Add some new symbols to the docs

2006-11-20 16:44:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/ffmpegcolorspace.c:
	  (ffmpegcolorspace_suite):
	  Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
	  (for now not for valgrinding though, since it takes too long).

2006-11-20 15:01:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_pixfmt_to_caps):
	  Fix RGBA32 caps. Fixes #357038.

2006-11-20 12:20:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/mixertrack.h:
	  Add FIXME so we can add some padding here in 0.11

2006-11-19 17:07:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  Fix GstBaseRTPAudioPayload structure so the whole GObject
	  inheritance business actually works (parent class instance structure
	  must always come first in the derived class instance structure).

2006-11-16 14:35:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
	  Original commit message from CVS:
	  * gst/videotestsrc/Makefile.am:
	  * tests/check/Makefile.am:
	  Make sure our checks and the videotestsrc plugin link against the
	  local uninstalled gst libs and not any installed gst libs that
	  might happen to exist as well.
	  * tests/check/elements/adder.c: (message_received),
	  (test_event_message_received), (test_play_twice_message_received):
	  * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
	  Fix compiler warnings when compiling against core with disabled
	  debugging system.

2006-11-16 12:55:08 +0000  Michael Smith <msmith@xiph.org>

	  gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	  (gst_audio_rate_sink_event), (gst_audio_rate_chain):
	  Fix audiorate, so that it accurately sets offsets and timestamps.
	  Doesn't change the fundamental algorithmic decisions; so should be
	  safe.
	  * tests/check/Makefile.am:
	  Enable audiorate test now that it passes.

2006-11-15 10:05:33 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
	  clear xv when going to NULL, remove // commented non-existant proto
	  * tests/examples/seek/seek.c: (main):
	  add missing tooltip description for scrub and play_scrub

2006-11-14 23:34:19 +0000  David Schleef <ds@schleef.org>

	  configure.ac: Bump liboil requirement to 0.3.8.
	  Original commit message from CVS:
	  * configure.ac:
	  Bump liboil requirement to 0.3.8.
	  * gst-libs/gst/riff/riff-media.c:
	  Add Dirac fourcc.
	  * gst/videoscale/vs_image.h:
	  * gst/videoscale/vs_scanline.h:
	  Use liboil's stdint.h.
	  * gst/videotestsrc/videotestsrc.c:
	  Remove liboil related ifdef's, since they aren't needed now, and
	  won't work with future versions.

2006-11-14 23:08:38 +0000  David Schleef <ds@schleef.org>

	  gst/videoscale/: Add a 4-tap image scaler.  Theoretically looks much prettier.
	  Original commit message from CVS:
	  * gst/videoscale/Makefile.am:
	  * gst/videoscale/gstvideoscale.c:
	  * gst/videoscale/gstvideoscale.h:
	  * gst/videoscale/vs_4tap.c:
	  * gst/videoscale/vs_4tap.h:
	  * gst/videoscale/vs_image.c:
	  * gst/videoscale/vs_image.h:
	  * gst/videoscale/vs_scanline.c:
	  * gst/videoscale/vs_scanline.h:
	  Add a 4-tap image scaler.  Theoretically looks much prettier.
	  The tap calculation could use some improvement.

2006-11-14 11:54:14 +0000  Jan David Mol <j.j.d.mol@tudelft.nl>

	  Various gsize and gssize printf fixes. Fixes #372507.
	  Original commit message from CVS:
	  Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
	  (gst_riff_parse_strf_iavs):
	  * gst/subparse/gstsubparse.c: (convert_encoding):
	  * gst/tcp/gstmultifdsink.c:
	  (gst_multi_fd_sink_handle_client_write):
	  * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
	  (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
	  (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
	  * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
	  * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
	  (gst_ximagesink_ximage_new):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
	  Various gsize and gssize printf fixes. Fixes #372507.

2006-11-13 18:14:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
	  (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
	  (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
	  (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
	  (vorbis_dec_chain_forward), (vorbis_dec_chain):
	  * ext/vorbis/vorbisdec.h:
	  First stab at vorbis reverse playback.

2006-11-13 17:30:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_event), (gst_base_audio_sink_render):
	  * gst-libs/gst/audio/gstbaseaudiosink.h:
	  Make the clock sync code more accurate wrt resampling and playback
	  at different rates.
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Use better algorithm to interpolate sample rates.

2006-11-13 15:31:01 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggdemux.c: Improve a debug line slightly.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
	  Improve a debug line slightly.
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
	  Call gst_riff_init() in plugin_init, to avoid getting errors from
	  the debug system (unrelated changes to another plugin made this turn
	  up; not sure why).

2006-11-10 19:20:21 +0000  Sergey Scobich <sergery.scobich@gmail.com>

	  win32/common/libgsttag.def: Add missing symbol (#366492).
	  Original commit message from CVS:
	  Patch by: Sergey Scobich  <sergery.scobich at gmail com>
	  * win32/common/libgsttag.def:
	  Add missing symbol (#366492).

2006-11-10 00:52:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gststreamselector.c: Don't unref a NULL pad.
	  Original commit message from CVS:
	  * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
	  Don't unref a NULL pad.

2006-11-09 00:50:00 +0000  Christian Schaller <uraeus@gnome.org>

	  ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
	  (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
	  (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
	  (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
	  (gst_ogg_demux_loop):
	  Implement first stab at reverse playback.

2006-11-07 07:22:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	  (gst_riff_create_video_template_caps):
	  add h263/h264 variants to the caps, Fixes #363118

2006-11-06 18:24:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
	  * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
	  Use g_strerror instead of strerror so we get UTF-8.

2006-11-04 07:25:58 +0000  David Schleef <ds@schleef.org>

	  ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c:
	  * ext/ogg/gstoggmux.c:
	  Add/remove KW-DIRAC header here, since it is ogg-specific.

2006-11-03 15:44:31 +0000  Michael Smith <msmith@xiph.org>

	  gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
	  Recognise more mpeg4 elementary video streams.

2006-11-02 17:26:03 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
	  Lower the probability of mp3 typefinding functions if we don't find a
	  valid mp3 header at the start of the file.
	  Closes #369482

2006-11-02 15:06:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
	  Original commit message from CVS:
	  * ext/theora/gsttheoradec.h:
	  * ext/theora/theoradec.c: (gst_theora_dec_init),
	  (theora_dec_sink_event), (theora_dec_chain_forward),
	  (theora_dec_flush_decode), (theora_dec_chain_reverse),
	  (theora_dec_chain):
	  Document and partially implement an algorithm for doing reverse playback
	  of theora video.

2006-11-02 14:18:45 +0000  Sergey Scobich <sergey.scobich@gmail.com>

	  win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
	  Original commit message from CVS:
	  Patch by: Sergey Scobich  <sergey.scobich at gmail com>
	  * win32/common/config.h:
	  * win32/common/interfaces-enumtypes.c:
	  * win32/common/libgsttag.def:
	  * win32/vs8/gst-plugins-base.sln:
	  * win32/vs8/libgstaudioresample.vcproj:
	  * win32/vs8/libgstinterfaces.vcproj:
	  * win32/vs8/libgstogg.vcproj:
	  * win32/vs8/libgstriff.vcproj:
	  * win32/vs8/libgsttag.vcproj:
	  * win32/vs8/libgsttheora.vcproj:
	  * win32/vs8/libgstvideoscale.vcproj:
	  * win32/vs8/libgstvorbis.vcproj:
	  Misc. VS8 build fixes: fix syntax in config.h, add missing entries
	  to libgsttag.def; add missing dependencies for some vs8 projects;
	  re-arrange placement of .def files in vs8 projects (#366334).

2006-11-01 14:08:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstogg.c: Remove unused variable.
	  Original commit message from CVS:
	  * ext/ogg/gstogg.c:
	  Remove unused variable.
	  * ext/ogg/gstoggdemux.c:
	  Fix Wim's surname in plugin description.

2006-10-31 15:05:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
	  Original commit message from CVS:
	  * gst-plugins-base.spec.in:
	  spec new .h file. Fixes #368310.

2006-10-31 14:19:07 +0000  Michael Smith <msmith@xiph.org>

	  gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
	  (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
	  (gst_multi_fd_sink_get_stats),
	  (gst_multi_fd_sink_remove_client_link),
	  (gst_multi_fd_sink_queue_buffer),
	  (gst_multi_fd_sink_handle_clients):
	  * gst/tcp/gstmultifdsink.h:
	  Make using the remove or clear signals threadsafe.
	  Make calling get-stats with an invalid fd not segfault.
	  Fixes 368273.

2006-10-31 10:49:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/: Fix and activate base audio payloader.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/Makefile.am:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  (gst_base_rtp_audio_payload_init):
	  Fix and activate base audio payloader.

2006-10-28 17:22:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
	  (plugin_init):
	  Add typefinder for QuickTime Image Files (see #366156).

2006-10-28 16:00:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
	  Another typo fix (#366212).

2006-10-27 17:13:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c: (volume_transform_ip):
	  Use stream time to synchronize volume property instead of rather random
	  timestamps. This is needed when gnonlin does its time shifting.

2006-10-27 16:46:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ChangeLog:
	  I'm too lazy to comment this
	  Original commit message from CVS:
	  *** empty log message ***

2006-10-27 16:45:30 +0000  Mark Nauwelaerts <manauw@skynet.be>

	  ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
	  Original commit message from CVS:
	  Patch by: Mark Nauwelaerts <manauw at skynet dot be>
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
	  Remove the pad from the element in release_pad.

2006-10-27 11:57:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
	  Explicitly create our custom buffer classes at a thread-safe
	  location as well, since g_type_class_ref() doesn't seem to be
	  entirely thread-safe either (#365501; also see #349410).

2006-10-26 10:49:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-read.c: If strings in INFO chunk are not UTF-8, do something similar to what we do for ID3v1 t...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
	  (gst_riff_parse_info):
	  If strings in INFO chunk are not UTF-8, do something similar to
	  what we do for ID3v1 tags: check a number of environment variables
	  (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
	  character sets to try, otherwise try the current locale and/or fall
	  back on ISO-8859-1. Fixes #360552.

2006-10-23 12:46:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  (gst_video_test_src_pattern_get_type),
	  (gst_video_test_src_set_pattern):
	  * gst/videotestsrc/gstvideotestsrc.h:
	  * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
	  (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
	  (gst_video_test_src_checkers8):
	  * gst/videotestsrc/videotestsrc.h:
	  Add a bunch of exciting new checkers patterns.

2006-10-23 12:06:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
	  Original commit message from CVS:
	  * gst/subparse/Makefile.am:
	  * gst/subparse/gstsubparse.c:
	  (gst_sub_parse_data_format_autodetect),
	  (gst_sub_parse_format_autodetect), (handle_buffer),
	  (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
	  * gst/subparse/gstsubparse.h:
	  * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
	  (parse_tmplayer):
	  * gst/subparse/tmplayerparse.h:
	  Add support for TMPlayer-type subtitles (#362845).
	  * tests/check/elements/subparse.c: (test_tmplayer_do_test),
	  (GST_START_TEST), (subparse_suite):
	  Add some basic unit tests for the above.

2006-10-23 11:47:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
	  Original commit message from CVS:
	  * tests/check/elements/audiorate.c: (test_injector_base_init),
	  (test_injector_class_init), (test_injector_chain),
	  (test_injector_init), (probe_cb), (do_perfect_stream_test),
	  (GST_START_TEST), (audiorate_suite):
	  More tests for audiorate: inject buffers to check behaviour when
	  buffers overlap.

2006-10-21 16:39:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
	  (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
	  Add some basic unit tests for audiorate. Disabled at the moment
	  since it doesn't pass yet (see bug #363119).

2006-10-20 17:02:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
	  (parse_subrip), (handle_buffer):
	  Add missing closing tags for markup and fix broken markup,
	  otherwise pango won't render anything (fixes #357531). Also,
	  make sure the text we send out is always NUL-terminated
	  (better safe than sorry etc.).
	  * tests/check/elements/subparse.c: (test_srt_do_test),
	  (test_srt):
	  Some more tests for .srt incl. tests for the above stuff.

2006-10-20 13:56:55 +0000  Stefan Kost <ensonic@users.sf.net>

	  sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
	  Original commit message from CVS:
	  2006-10-20  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
	  Patch by: Stefan Kost  <ensonic@users.sf.net>
	  Try to redraw borders only when needed. Apparently this consumes
	  resources on small devices... :-O (#363607)

2006-10-20 13:54:19 +0000  Michael Smith <msmith@xiph.org>

	  gst/tcp/gstmultifdsink.c: If caps change, then update the client's idea of the caps so that we don't end up re-sendin...
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c:
	  (gst_multi_fd_sink_client_queue_buffer):
	  If caps change, then update the client's idea of the caps so that we
	  don't end up re-sending streamheaders for every single buffer after
	  the caps change.

2006-10-20 12:31:02 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
	  Original commit message from CVS:
	  * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
	  (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
	  Set caps on pushed buffers; fix up refcounting of caps objects.

2006-10-19 14:09:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
	  (plugin_init):
	  Typefind mmsh header data packet to application/x-mmsh (#362625).

2006-10-19 09:17:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add very simple unit test for subparse.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/subparse.c: (buffer_from_static_string),
	  (setup_subparse), (teardown_subparse), (test_srt_do_test),
	  (GST_START_TEST), (subparse_suite):
	  Add very simple unit test for subparse.

2006-10-19 09:00:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
	  (parse_subrip):
	  Strip trailing newlines from subtitle text output.

2006-10-18 18:40:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	  (gst_sub_parse_change_state):
	  Fix memleak; clear subparse->textbuf n state change function.

2006-10-18 15:13:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c:
	  (gst_sub_parse_data_format_autodetect):
	  Don't require subrip (.srt) files to start with a chunk number of 1.

2006-10-18 13:42:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_event), (gst_base_audio_sink_render):
	  * gst-libs/gst/audio/gstbaseaudiosink.h:
	  Extract rate from the NEWSEGMENT event.
	  Use commit_full to also take rate adjustment into account when writing
	  samples to the ringbuffer.
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
	  (gst_ring_buffer_read):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Added _commit_full() to also take rate into account.
	  Use simple interpolation algorithm to resample audio.
	  API: gst_ring_buffer_commit_full()
	  * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
	  * tests/examples/seek/seek.c: (segment_done):
	  Don't try to seek with 0.0 rate, just pause instead.
	  Remove bogus debug line.

2006-10-18 12:57:54 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
	  (setup_source):
	  Catch async errors when starting up the subtitle bin, so we can
	  stop waiting and continue with the main film instead of hanging
	  forever. Fixes #339366.
	  * tests/check/elements/playbin.c: (playbin_suite):
	  Enable unit test for the above.

2006-10-18 09:53:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/playbin.c: (GST_START_TEST),
	  (gst_red_video_src_uri_get_type),
	  (gst_red_video_src_uri_get_protocols),
	  (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
	  (gst_red_video_src_uri_handler_init),
	  (gst_red_video_src_init_type), (gst_red_video_src_base_init),
	  (gst_red_video_src_create), (gst_red_video_src_class_init),
	  (gst_red_video_src_init), (plugin_init), (playbin_suite):
	  Some small and basic unit tests for playbin; not very useful yet,
	  but at least a start.

2006-10-18 09:46:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: The old pad activation spiel.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (setup_sinks):
	  The old pad activation spiel.

2006-10-18 09:31:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_source):
	  Don't hang forever if the subbin already fails to start up in
	  the state change to PAUSED (#339366).

2006-10-17 17:17:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
	  (gst_tuner_set_channel), (gst_tuner_get_channel),
	  (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
	  (gst_tuner_set_frequency), (gst_tuner_get_frequency),
	  (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
	  (gst_tuner_find_channel_by_name):
	  Fix some function guards, add some more function guards.

2006-10-17 11:34:32 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
	  (remove_element_chain):
	  Don't return a pad from get_our_ghost_pad unless it is actually the
	  one we want.
	  Change a cast in remove_element_chain slightly.

2006-10-13 15:20:29 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
	  Original commit message from CVS:
	  2006-10-13  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (do_seek), (start_seek),
	  (rate_spinbutton_changed_cb), (segment_done),
	  (msg_state_changed):
	  Segment seeking needs to use the rate and set stop to -1.

2006-10-13 14:15:42 +0000  Ville Syrjala <ville.syrjala@movial.fi>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_setcaps):
	  Don't crash when ringbuffer is not yet created.
	  Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
	  Fixes #361634.
	  * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_request_new_pad):
	  Activate pads befre adding them to running elements.

2006-10-13 11:25:10 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
	  Original commit message from CVS:
	  2006-10-13  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (do_seek), (start_seek),
	  (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
	  scale
	  updater when we start grabing the slider. Don't wait for the
	  pipeline to be PAUSED.

2006-10-13 08:57:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
	  (gst_mixer_set_volume), (gst_mixer_get_volume),
	  (gst_mixer_set_mute), (gst_mixer_set_option),
	  (gst_mixer_get_option), (gst_mixer_mute_toggled),
	  (gst_mixer_record_toggled), (gst_mixer_volume_changed),
	  (gst_mixer_option_changed):
	  Guard mixer interface functions against bogus arguments.

2006-10-12 19:39:07 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
	  Original commit message from CVS:
	  2006-10-12  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (do_seek), (start_seek),
	  (stop_seek),
	  (play_cb), (pause_cb), (stop_cb),
	  (rate_spinbutton_changed_cb),
	  (msg_state_changed), (main): Use state-changed messages to
	  trigger
	  start/stop of scale update timer. Indeed the scale slider was
	  jumping here and there because the update timer was activated
	  before seek completed. This fixes instant applying of rate
	  changes
	  by pressing the spinbutton like a crazy man !

2006-10-12 19:09:06 +0000  Sebastien Cote <sebas642@yahoo.ca>

	  gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
	  Original commit message from CVS:
	  Patch by: Sebastien Cote  <sebas642 at yahoo.ca>
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
	  (gst_basertppayload_finalize):
	  Fix two small memory leaks (#361456).

2006-10-10 18:56:01 +0000  Julien Moutte <julien@moutte.net>

	  tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
	  Original commit message from CVS:
	  2006-10-10  Julien MOUTTE  <julien@moutte.net>
	  * tests/examples/seek/seek.c: (do_seek),
	  (rate_spinbutton_changed_cb): When changing spinbutton we try
	  to change the rate on the fly.

2006-10-10 16:50:06 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/riff/: Add WMS caps.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-ids.h:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	  (gst_riff_create_audio_template_caps):
	  Add WMS caps.

2006-10-10 12:49:03 +0000  Josep Torre Valles <josep@fluendo.com>

	  ext/gnomevfs/: Fix URI interface implementation return type.
	  Original commit message from CVS:
	  2006-10-10  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  Patch by: Josep Torre Valles <josep@fluendo.com>
	  * ext/gnomevfs/gstgnomevfssink.c:
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  Fix URI interface implementation return type.
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
	  Fix what looks like a copy/paste issue when assigning values.
	  * gst-libs/gst/audio/gstaudiofiltertemplate.c:
	  (gst_audio_filter_template_get_type):
	  Cast to prevent Forte warnings.
	  * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
	  Fix URI interface implementation return type.
	  gst_pad_query_position requires a signed integer pointer as
	  3rd parameter, GstClockTime is unsigned.
	  * gst/audioconvert/audioconvert.c:
	  Fix integer overflow when treated as signed.
	  * gst/audioresample/resample.c: (resample_add_input_data):
	  Cast to prevent warnings on Forte.
	  * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
	  Fix integer overflow when treated as signed.
	  * gst/ffmpegcolorspace/imgconvert_template.h:
	  Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
	  * gst/playback/gstdecodebin.c: (queue_filled_cb),
	  (cleanup_decodebin):
	  Who initialises a guint to -1!
	  Cast function pointers to prevent warnings on Forte.
	  * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
	  (queue_threshold_reached):
	  Cast function pointers correctly to prevent warnings on Forte.
	  * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
	  Cast function pointers correctly to prevent warnings on Forte.
	  * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
	  Obvious change to unsigned, 0xEF > max signed char.
	  * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
	  GstClockTime is unsigned, initialise correctly.
	  * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
	  Cast so pointer arithemetic doesn't cause warnings on Forte.
	  * gst/videorate/gstvideorate.c:
	  Use correct return value.
	  * tests/examples/seek/scrubby.c:
	  GstClockTime is unsigned, initialise correctly.

2006-10-10 11:20:03 +0000  Ferenc Gerlits <fgerlits@gmail.com>

	  gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
	  Original commit message from CVS:
	  Patch by: Ferenc Gerlits  <fgerlits at gmail com>
	  * gst/typefind/gsttypefindfunctions.c:
	  Recognise XML files and XML-like files shorter than 256 bytes as
	  well (fixes #359237).

2006-10-09 15:01:30 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* common:
	* gst/typefind/gsttypefindfunctions.c:
	  Added typefind functions to video/x-nuv media.
	  Original commit message from CVS:
	  Added typefind functions to video/x-nuv media.

2006-10-08 16:59:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/xoverlay.c:
	  (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
	  Some more guards against invalid input.

2006-10-07 18:35:39 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.c: Useless goto.
	  Original commit message from CVS:
	  2006-10-07  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
	  Useless goto.
	  * tests/examples/seek/seek.c: (do_seek),
	  (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
	  seek example to experiment with rates != 1.0 (reverse playback
	  !)

2006-10-06 19:20:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/xoverlay.c:
	  Unref message in doc-example (spotted by Robert McQueen)

2006-10-06 17:04:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/typefind/gsttypefindfunctions.c: printf fix.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
	  (mpeg1_parse_header), (mpeg1_sys_type_find):
	  printf fix.

2006-10-06 14:37:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/: Activate dynamic pads before adding them to the element.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
	  (close_pad_link):
	  * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
	  Activate dynamic pads before adding them to the element.

2006-10-06 14:04:53 +0000  Michael Smith <msmith@xiph.org>

	  gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
	  Original commit message from CVS:
	  * gst-libs/gst/floatcast/floatcast.h:
	  Fix obviously-bogus macros; use the correct types.

2006-10-06 13:34:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_change_state):
	  Also call parent state change function to activate pads.
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
	  (mpeg1_parse_header), (mpeg1_sys_type_find):
	  Add some more debug info in mpeg typefinding.

2006-10-06 12:57:10 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_dec_chain):
	  Zero byte theora packets are valid and well-defined; don't warn on
	  them.

2006-10-06 10:04:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
	  (gst_multi_fd_sink_get_stats), (find_limits),
	  (gst_multi_fd_sink_queue_buffer):
	  API: add dropped_buffers to the get-stats GValueArray

2006-10-05 15:55:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Printf format fixes.
	  Original commit message from CVS:
	  * ext/alsa/gstalsadeviceprobe.c:
	  (gst_alsa_device_property_probe_get_values):
	  * ext/alsa/gstalsasink.c: (set_hwparams):
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
	  (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
	  (gst_ogg_mux_process_best_pad):
	  * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
	  (gst_ogg_parse_chain):
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
	  (gst_vorbis_enc_buffer_check_discontinuous):
	  * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_handle_track_seek):
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_push_full):
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	  * gst/audioresample/resample.c: (resample_input_pushthrough):
	  * gst/playback/gstplaybasebin.c: (queue_out_of_data):
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
	  (wavpack_type_find):
	  * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
	  * tests/check/elements/volume.c: (GST_START_TEST):
	  Printf format fixes.

2006-10-04 13:18:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
	  Original commit message from CVS:
	  * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
	  Fix a simple mistake (see the docs)
	  Fixes #359580

2006-10-04 13:15:36 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* win32/common/config.h:
	  bump version
	  Original commit message from CVS:
	  bump version

2006-10-03 12:11:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * docs/plugins/gst-plugins-base-plugins.args:
	  * docs/plugins/gst-plugins-base-plugins.hierarchy:
	  * docs/plugins/inspect/plugin-adder.xml:
	  * docs/plugins/inspect/plugin-alsa.xml:
	  * docs/plugins/inspect/plugin-audioconvert.xml:
	  * docs/plugins/inspect/plugin-audiorate.xml:
	  * docs/plugins/inspect/plugin-audioresample.xml:
	  * docs/plugins/inspect/plugin-audiotestsrc.xml:
	  * docs/plugins/inspect/plugin-cdparanoia.xml:
	  * docs/plugins/inspect/plugin-decodebin.xml:
	  * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	  * docs/plugins/inspect/plugin-gdp.xml:
	  * docs/plugins/inspect/plugin-gnomevfs.xml:
	  * docs/plugins/inspect/plugin-libvisual.xml:
	  * docs/plugins/inspect/plugin-ogg.xml:
	  * docs/plugins/inspect/plugin-pango.xml:
	  * docs/plugins/inspect/plugin-playbin.xml:
	  * docs/plugins/inspect/plugin-subparse.xml:
	  * docs/plugins/inspect/plugin-tcp.xml:
	  * docs/plugins/inspect/plugin-theora.xml:
	  * docs/plugins/inspect/plugin-typefindfunctions.xml:
	  * docs/plugins/inspect/plugin-video4linux.xml:
	  * docs/plugins/inspect/plugin-videorate.xml:
	  * docs/plugins/inspect/plugin-videoscale.xml:
	  * docs/plugins/inspect/plugin-videotestsrc.xml:
	  * docs/plugins/inspect/plugin-volume.xml:
	  * docs/plugins/inspect/plugin-vorbis.xml:
	  * docs/plugins/inspect/plugin-ximagesink.xml:
	  * docs/plugins/inspect/plugin-xvimagesink.xml:
	  Add vorbistag element to docs; update version numbers to 0.10.10.1.

2006-10-03 11:51:48 +0000  James Doc Livingston <doclivingston@gmail.com>

	  ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
	  Original commit message from CVS:
	  Patch by: James "Doc" Livingston <doclivingston at gmail com>
	  * ext/vorbis/Makefile.am:
	  * ext/vorbis/vorbis.c: (plugin_init):
	  * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
	  (vorbis_parse_parse_packet), (vorbis_parse_chain):
	  * ext/vorbis/vorbisparse.h:
	  * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
	  (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
	  (gst_vorbis_tag_parse_packet):
	  * ext/vorbis/vorbistag.h:
	  Add new vorbistag element which derives from vorbisparse
	  and is essentially the same as well, only that it implements
	  the GstTagSetter interface and can modify the stream's
	  vorbiscomment on the fly (#335635).
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/vorbistag.c: (setup_vorbistag),
	  (cleanup_vorbistag), (buffer_probe), (start_pipeline),
	  (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
	  (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
	  Add unit test for new vorbistag element.

2006-10-03 10:36:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
	  (vorbis_parse_push_headers), (vorbis_parse_chain):
	  Set BOS flag in packet structure to fix 'jump depends
	  on unitialized value' errors in valgrind; various minor
	  clean-ups.

2006-09-30 15:30:07 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstdecodebin.c: Fix typo in a debug statement.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (close_pad_link):
	  Fix typo in a debug statement.
	  * gst/playback/gstplaybasebin.c: (probe_triggered),
	  (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
	  (gen_source_element), (source_new_pad), (analyse_source),
	  (setup_source):
	  When handling no_more_pads in new_decoded_pad, make sure to treat
	  subtitle pads correctly. Fixes playback with subtitle files.
	  Move a recurring message to LOG level.
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
	  The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
	  which ends up as -1 when cast to an int. Make the logic handle the
	  max value as an unsigned mask and only change the colorkey when it's
	  a value we recognise.

2006-09-30 00:14:20 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  Removed empty * between paragraphs

2006-09-29 23:50:53 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/: Moved some documentation into .c file
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst-libs/gst/rtp/README:
	  Moved some documentation into .c file

2006-09-29 17:35:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Fix compilation.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (no_more_pads):
	  Fix compilation.

2006-09-29 16:04:05 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Remove g_print
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (new_caps):
	  Remove g_print
	  * gst/playback/gstplaybin.c:
	  Add some docs.

2006-09-29 15:16:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Re-enable cddabasesrc test to see if it works again
	  now.

2006-09-29 13:46:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_subtitle),
	  (gen_source_element):
	  Handle invalid URIs a bit more gracefully.

2006-09-29 12:54:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/pipelines/oggmux.c: Remove obsolete comment.
	  Original commit message from CVS:
	  * tests/check/pipelines/oggmux.c:
	  Remove obsolete comment.

2006-09-29 10:43:05 +0000  James Doc Livingston <doclivingston@gmail.com>

	  ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
	  (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
	  (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
	  (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
	  (gst_ogg_mux_collected):
	  Commit patch from James "Doc" Livingston, adds proper EOS handling
	  in oggmux. GStreamer can, for the first time ever, create a valid
	  Ogg file! Yay!
	  * tests/check/pipelines/oggmux.c: (check_chain_final_state),
	  (oggmux_suite):
	  Reenable tests now that they pass.

2006-09-29 08:20:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
	  Stop reading commands when EOF (we read 0) as well.

2006-09-28 15:29:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
	  (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
	  (find_dynamic), (unlinked), (close_link):
	  Implement delayed caps linking needed for element with a lot of
	  different caps on the src pads that get fixed at runtime.
	  Improve management of dynamic elements.
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
	  (group_destroy), (group_commit), (check_queue), (queue_overrun),
	  (gen_preroll_element), (remove_groups), (unknown_type),
	  (add_element_stream), (no_more_pads_full), (no_more_pads),
	  (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
	  (new_decoded_pad), (setup_subtitle), (array_has_value),
	  (gen_source_element), (source_new_pad), (has_all_raw_caps),
	  (analyse_source), (remove_decoders), (make_decoder),
	  (remove_source), (setup_source), (finish_source), (prepare_output),
	  (gst_play_base_bin_change_state):
	  * gst/playback/gstplaybasebin.h:
	  Use more _CAST instead of full type checking casts.
	  Small cleanups, plug some leaks.
	  Handle dynamic sources.
	  Add some helper functions to create lists of strings used for
	  blacklisting and other stuff.
	  Refactor some code dealing with analysing the source.
	  Re-enable sources without pads (like cd:// or other selfcontained
	  elements).

2006-09-28 15:08:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  When we have a timestamp, we can still perform clipping.
	  When we have no clock, we must play the sample ASAP.

2006-09-28 11:46:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  Set caps on outgoing buffers.
	  * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
	  (gst_video_rate_event), (gst_video_rate_chain):
	  * gst/videorate/gstvideorate.h:
	  Fix videorate some more. Fixes #357977

2006-09-28 11:34:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
	  Original commit message from CVS:
	  * tests/check/elements/adder.c: (adder_suite):
	  Don't set timeout to 6 seconds when we're running
	  in valgrind ... (and how is 6 seconds longer than
	  the default anyway?)

2006-09-28 10:49:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	  (gst_audio_rate_sink_event), (gst_audio_rate_convert),
	  (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
	  Keep sink and src segment to keep track of time and support more
	  input formats.
	  Fix bogus next_offset and run_time calculation, don't understand how
	  this could have worked before. Fixes #357976.
	  Remove some unneeded vars.

2006-09-28 09:41:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (remove_sinks):
	  Only remove visualisation from visbin if there is a visbin (or:
	  don't throw warnings when closing totem without playing a file).

2006-09-27 13:52:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  Add some more info in a WARNING.
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_create):
	  Handle PAUSE in create function, use new -core addition to
	  wait for playing. Fixes pausing and resuming capture from an
	  audiosrc.
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
	  (gst_ring_buffer_read):
	  Constify some more.
	  Caller supports interrupted reads now.

2006-09-27 13:29:49 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  add new header file to spec
	  Original commit message from CVS:
	  add new header file to spec

2006-09-27 12:55:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Another attempt to make the gen64 buildbot happy.

2006-09-27 11:58:17 +0000  Jonathan Matthew <jonathan@kaolin.wh9.net>

	  ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
	  Original commit message from CVS:
	  Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
	  * ext/libvisual/visual.c: (gst_visual_clear_actors),
	  (gst_visual_chain), (gst_visual_change_state):
	  Libvisual plugin was not passing audio data to libvisual 0.4.0
	  correctly. Fixes #357800

2006-09-27 11:31:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
	  Original commit message from CVS:
	  * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
	  Add timeout to _get_state() so we see which pipeline it is
	  that causes trouble on the gen64 build bot.

2006-09-27 11:06:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
	  (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  the source pad always uses fixed caps.

2006-09-27 11:05:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Added docs for the audio libs.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/audio/gstaudioclock.c:
	  * gst-libs/gst/audio/gstaudioclock.h:
	  * gst-libs/gst/audio/gstaudiosink.c:
	  * gst-libs/gst/audio/gstaudiosink.h:
	  * gst-libs/gst/audio/gstaudiosrc.c:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  * gst-libs/gst/audio/gstbaseaudiosink.h:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
	  * gst-libs/gst/audio/gstbaseaudiosrc.h:
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Added docs for the audio libs.

2006-09-27 10:59:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Temporarily disable test that fails on the bots for unknown reasons.

2006-09-27 00:13:29 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  Moved AudioCodecType into priv
	  Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes

2006-09-25 15:47:25 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
	  (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
	  (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
	  (new_pad):
	  Cleanups and small leak fixes.
	  Added Depayloaders to valid list of autopluggable elements.

2006-09-25 13:24:59 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	  (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
	  (gen_video_element), (gen_text_element), (gen_audio_element),
	  (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
	  (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
	  Detect NO_PREROLL state change returns and disable clock distribution to
	  the sinks so that sync is disabled.
	  Avoid some type checking and do simple casts instead.
	  Small cleanups, fix some FIXMEs.
	  Be more robust when linking user specified elements, catch an report
	  errors. Fixes #357404.
	  Fix some leaks in the error paths.

2006-09-25 12:55:04 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	* ChangeLog:
	  ChangeLog surgery for missing bug-number
	  Original commit message from CVS:
	  ChangeLog surgery for missing bug-number

2006-09-25 11:28:15 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/playback/test.c:
	  Fix compilation with uClibc and -Werror (#357591).

2006-09-25 10:21:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Parse dates that are followed by a time as well (#357532).
	  * tests/check/libs/tag.c: (test_vorbis_tags):
	  Add unit test for this.

2006-09-23 15:24:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: A few array const-ifications.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	  (gst_audio_convert_transform_caps):
	  * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
	  * gst/videotestsrc/videotestsrc.h:
	  A few array const-ifications.

2006-09-23 15:02:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: See if this makes the build bots happy.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  See if this makes the build bots happy.
	  * tests/check/libs/cddabasesrc.c:
	  UTF8-ise my name.

2006-09-23 14:30:53 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha <ganadist at chollian dot net>
	  * gst/subparse/samiparse.c: (handle_start_font),
	  (fix_invalid_entities):
	  More case-insensitivity for certain tags; recognise entities with
	  decimal codes as special entities as well (#357330).

2006-09-23 13:32:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
	  Original commit message from CVS:
	  * gst-libs/gst/Makefile.am:
	  Need to build tag directory before cdda.

2006-09-23 13:21:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/cdda/Makefile.am:
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_base_init):
	  * gst-libs/gst/cdda/gstcddabasesrc.h:
	  * gst-libs/gst/tag/tag.h:
	  * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
	  (gst_tag_register_musicbrainz_tags):
	  Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
	  depend on libgsttag. This is required so we can extract/read tags like
	  DISCID without depending on libgstcddabasesrc (which used to register
	  them).
	  * gst-libs/gst/tag/gstvorbistag.c:
	  Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
	  tags (also see #347848).
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
	  Log vorbis comments we are actually writing. Const-ify array.

2006-09-23 08:53:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gen_preroll_element):
	  Improve buffering a bit by avoiding a deadlock because we cannot assume
	  the underrun is always called.

2006-09-23 08:51:14 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha <ganadist at chollian dot net>
	  * gst-libs/gst/riff/riff-ids.h:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	  (gst_riff_create_audio_template_caps):
	  Added MPEG-4 AAC and id and caps. Fixes #357289
	  Added WMA9 Lossless id.

2006-09-22 14:50:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  Fix misleading docs addition.
	  * tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  Get rid of compiler warning the right way.

2006-09-22 14:13:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_finalize),
	  (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
	  (gst_base_rtp_depayload_push_full),
	  (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
	  (gst_base_rtp_depayload_process),
	  (gst_base_rtp_depayload_set_gst_timestamp),
	  (gst_base_rtp_depayload_queue_release):
	  * gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Small cleanups.
	  Fix some leaks.
	  Refactored the process method and added methods to push from the process
	  vmethod.
	  Use _scale functions.
	  API: gst_base_rtp_depayload_push_ts
	  API: gst_base_rtp_depayload_push
	  * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
	  timestamps are uint.

2006-09-22 11:59:00 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/xoverlay.c:
	  Remove unused statement from doc example.

2006-09-22 09:52:21 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/videorate/gstvideorate.c:
	  update docs
	  Original commit message from CVS:
	  update docs

2006-09-21 13:49:47 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/videoorientation.c:
	  (gst_video_orientation_iface_init),
	  (gst_video_orientation_get_hflip),
	  (gst_video_orientation_get_vflip),
	  (gst_video_orientation_get_hcenter),
	  (gst_video_orientation_get_vcenter),
	  (gst_video_orientation_set_hflip),
	  (gst_video_orientation_set_vflip),
	  (gst_video_orientation_set_hcenter),
	  (gst_video_orientation_set_vcenter):
	  Add since tags to new API docs, ChangeLog surgery (forgot API keyword
	  in ChangeLog)

2006-09-21 09:27:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: but disable for now since it doesn't pass (something wrong with
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
	  (create_rgb_conversions), (rgb_conversion_free),
	  (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
	  (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
	  Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
	  but disable for now since it doesn't pass (something wrong with
	  RGBA somewhere).

2006-09-21 07:01:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (group_commit),
	  (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
	  (queue_out_of_data), (gen_preroll_element),
	  (preroll_remove_overrun), (probe_triggered):
	  Refactor handling of overrun detection.
	  Separate handling of group completion and deadlock detection when doing
	  network buffering. This should fix some deadlocks that were not detected
	  because the group was completed.
	  Add more comments, improve debugging.

2006-09-21 05:31:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/: Some more compilation fixes.
	  Original commit message from CVS:
	  * tests/check/elements/gdpdepay.c: (GST_START_TEST):
	  * tests/check/libs/audio.c:
	  Some more compilation fixes.

2006-09-21 05:12:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
	  (gst_ring_buffer_read):
	  Early morning compilation fix.

2006-09-20 18:09:43 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  bump nano
	  Original commit message from CVS:
	  bump nano

2006-09-20 17:04:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/: Fix some warnings.
	  Original commit message from CVS:
	  * tests/check/elements/gdpdepay.c: (GST_START_TEST):
	  * tests/check/elements/multifdsink.c: (GST_START_TEST):
	  * tests/check/elements/videorate.c: (GST_START_TEST):
	  * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	  * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
	  Fix some warnings.

2006-09-20 10:59:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	  (gst_xvimagesink_get_times):
	  change colorkey behaviour back according to #354773 comment 6/7

2006-09-20 10:42:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: remove junk
	  Original commit message from CVS:
	  ChangeLog surgery: remove junk

2006-09-19 11:31:06 +0000  Michael Smith <msmith@xiph.org>

	  gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	  (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
	  (gst_multi_fd_sink_recover_client),
	  (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
	  (gst_multi_fd_sink_get_property):
	  * gst/tcp/gstmultifdsink.h:
	  Implement stubbed out properties unit-type, units-soft-max,
	  units-max, to allow specifying maximum sizes in units other than
	  buffers.
	  Fixes #355935

2006-09-19 10:23:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	  (gst_riff_create_audio_template_caps):
	  Reorder the audio formats a bit for clarity.
	  Detect and create caps for MSGSM and MSN (WAV49).
	  Fixes #356596.
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
	  (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
	  Small cleanups, move error handling out of normal flow for clarity.

2006-09-18 15:59:39 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add new interface to control video orientation (fixes #354908)
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs.types:
	  * gst-libs/gst/interfaces/Makefile.am:
	  * gst-libs/gst/interfaces/videoorientation.c:
	  (gst_video_orientation_get_type),
	  (gst_video_orientation_iface_init),
	  (gst_video_orientation_get_hflip),
	  (gst_video_orientation_get_vflip),
	  (gst_video_orientation_get_hcenter),
	  (gst_video_orientation_get_vcenter),
	  (gst_video_orientation_set_hflip),
	  (gst_video_orientation_set_vflip),
	  (gst_video_orientation_set_hcenter),
	  (gst_video_orientation_set_vcenter):
	  * gst-libs/gst/interfaces/videoorientation.h:
	  Add new interface to control video orientation (fixes #354908)

2006-09-18 15:48:01 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  Use G_UNLIKELY in _create and log one more detail.
	  (gst_video_test_src_get_times), (gst_video_test_src_create):
	  * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
	  Use gst_util_uint64_scale_int in _get_times().

2006-09-18 15:00:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
	  Give better warning message (add object and detail).

2006-09-18 14:42:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
	  (gst_xvimagesink_get_times):
	  xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
	  #354773), use gst_util_uint64_scale_int in _get_times()

2006-09-18 14:21:45 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
	  Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
	  always true, leading to dropping all timestamps.

2006-09-18 11:40:14 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/libvisual/visual.c: update to work also with libvisual 0.4 API
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_vis_src_negotiate),
	  (gst_visual_chain), (gst_visual_change_state):
	  update to work also with libvisual 0.4 API
	  * tools/gst-launch-ext.1.in:
	  * tools/gst-visualise.1.in:
	  remove references to old man-pages
	  * tests/examples/seek/seek.c: (main):
	  add real meadi-buttons, add tool-tips for the seek-options, arrange
	  seek options in a table

2006-09-18 10:57:28 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
	  (gst_ogg_mux_push_buffer):
	  Don't generate out-of-order timestamps from oggmux, instead clamp
	  output timestamps to be >= the previously output ts.
	  Fixes #355595

2006-09-18 10:18:22 +0000  Michael Smith <msmith@xiph.org>

	  gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	  (gst_multi_fd_sink_class_init):
	  Updates, fixes, and typo corrections for multifdsink. No functional
	  changes.

2006-09-17 21:58:06 +0000  Michael Smith <msmith@xiph.org>

	  gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
	  Don't crash on truncated files - check that we got an 8 byte buffer
	  before trying to memcmp it.

2006-09-17 20:32:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (get_active_source):
	  Make stream-switching appear instant to the application
	  (ie. make sure that a g_object_get on 'current-foo' returns
	  the stream previously set with g_object_set(). Totem needs
	  this to update stream-related meta-info (like audio-codec)
	  correctly when switching streams.

2006-09-17 20:14:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
	  (gst_alsa_mixer_ensure_track_list):
	  Try harder to guess which mixer track is the master mixer
	  track (instead of just taking the first one that has a pvolume).
	  Fixes #342228.

2006-09-17 11:24:21 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
	  (gst_audio_convert_transform_caps):
	  Get structure-name just once.

2006-09-16 22:30:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/: Fix big batch of compiler warnings.
	  Original commit message from CVS:
	  * tests/check/elements/audioresample.c: (GST_START_TEST):
	  * tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  * tests/check/elements/volume.c: (GST_START_TEST):
	  * tests/check/elements/vorbisdec.c: (GST_START_TEST):
	  * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
	  (test_pipeline), (GST_START_TEST):
	  * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
	  * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
	  Fix big batch of compiler warnings.

2006-09-16 21:54:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  Add docs about icydemux usage in connection with gnomevfssrc
	  * ext/libvisual/visual.c:
	  * ext/ogg/gstoggaviparse.c:
	  * ext/ogg/gstoggdemux.c:
	  * ext/ogg/gstoggmux.c:
	  * ext/ogg/gstoggparse.c:
	  * gst-libs/gst/audio/gstaudiofiltertemplate.c:
	  * gst-libs/gst/audio/gstaudiosink.c:
	  * gst-libs/gst/audio/gstaudiosrc.c:
	  * gst/audiorate/gstaudiorate.c:
	  More G_OBJECT macro fixing.
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  Fix wrong info in header due to copy & paste

2006-09-15 14:53:44 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
	  (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
	  (gst_base_audio_src_create), (gst_base_audio_src_change_state):
	  Do the delay calculation in the source/sink base classes as this is
	  specific for the capture/playback mode.
	  Try to fixate a bit better, like round depth up to a multiple of 8
	  bigger than width.
	  Handle underruns correctly by marking DISCONT on buffers and adjusting
	  timestamps to handle the gap.
	  Set offset/offset_end correctly on buffers.
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
	  (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
	  (gst_ring_buffer_read):
	  Remove resync and underrun recovery from the ringbuffer.
	  Fix ringbuffer read code on under/overrun.

2006-09-15 11:17:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	  (gst_play_base_bin_init), (fill_buffer), (check_queue),
	  (queue_threshold_reached), (gst_play_base_bin_set_property),
	  (gst_play_base_bin_get_property):
	  * gst/playback/gstplaybasebin.h:
	  Don't use a 0 low watermark when buffering, it is catching starvation
	  way too late. Instead, use a 3 second queue with 30 and 95
	  percent low/high watermarks.
	  Added queue-min-threshold property to configure low watermark.
	  Use new _buffering message API.
	  Make queue_threshold variable big enough to store a uint64 time value.
	  API: playbin::queue-min-threshold property.

2006-09-15 09:13:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  configure.ac: We require 0.10.10.1 now because of _wait_preroll().
	  Original commit message from CVS:
	  * configure.ac:
	  We require 0.10.10.1 now because of _wait_preroll().
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  Use gst_base_sink_wait_preroll().

2006-09-15 09:09:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/: Use DEBUG_OBJECT more.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
	  * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
	  Use DEBUG_OBJECT more.

=== release 0.10.10 ===

2006-09-14 20:09:19 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* common:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gdp.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* ext/theora/theoraparse.c:
	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* gst/playback/gstplaybin.c:
	* tests/check/Makefile.am:
	* win32/common/config.h:
	  releasing 0.10.10
	  Original commit message from CVS:
	  releasing 0.10.10

2006-09-09 16:08:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	* win32/common/config.h:
	  second prerelease
	  Original commit message from CVS:
	  second prerelease

2006-09-07 19:01:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	  update bug in changelog
	  Original commit message from CVS:
	  update bug in changelog

2006-09-07 19:00:33 +0000  Michael Smith <msmith@fluendo.com>

	  Fix implementation of sync-method 'next-keyframe'
	  Original commit message from CVS:
	  patch by: Michael Smith <msmith at fluendo dot com>
	  * gst/tcp/gstmultifdsink.c: (is_sync_frame),
	  (gst_multi_fd_sink_client_queue_buffer),
	  (gst_multi_fd_sink_new_client):
	  * tests/check/elements/multifdsink.c: (GST_START_TEST),
	  (multifdsink_suite):
	  Fix implementation of sync-method 'next-keyframe'

2006-09-07 15:00:08 +0000  Wim Taymans <wim@fluendo.com>

	  ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
	  Original commit message from CVS:
	  patch by: Wim Taymans <wim at fluendo dot com>
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
	  This patch removes the RANDOM flag that was incorrectly introduced with
	  revision 1.91.  Fixes #354590

2006-09-07 14:56:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	* win32/common/config.h:
	  first prerelease
	  Original commit message from CVS:
	  first prerelease

2006-09-07 14:56:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  update po files
	  Original commit message from CVS:
	  update po files

2006-09-05 09:12:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Random variation in Makefile line to see if it makes the
	  gen64-base-full bot any happier.

2006-09-04 19:04:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
	  Original commit message from CVS:
	  * tests/check/pipelines/oggmux.c: (oggmux_suite):
	  Disable test that fails at the moment (killed after timeout).

2006-09-04 18:19:06 +0000  James Livingston <doclivingston@gmail.com>

	  tests/check/: Add simple unit test for oggmux from #337026 with checking for the
	  Original commit message from CVS:
	  Patch by: James Livingston  <doclivingston at gmail.com>
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/.cvsignore:
	  * tests/check/pipelines/oggmux.c: (get_page_codec),
	  (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
	  (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
	  (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
	  (test_theora_vorbis), (oggmux_suite):
	  Add simple unit test for oggmux from #337026 with checking for the
	  EOS flags disabled for the time being.

2006-09-04 09:13:01 +0000  Alessandro Dessina <alessandro@nnva.org>

	  ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
	  Original commit message from CVS:
	  patch by: Alessandro Dessina <alessandro nnva org>
	  * ext/ogg/gstoggmux.c:
	  Add cmml caps to oggmux. Fixes #353912

2006-09-02 13:20:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
	  Original commit message from CVS:
	  * tests/check/elements/videotestsrc.c: (check_rgb_buf):
	  Returning a return value often helps. In this case, we
	  don't need the return value anyway, so just get rid of it.
	  Should make build bots much happier.

2006-09-02 12:59:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
	  Original commit message from CVS:
	  * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
	  (paint_get_structure), (gst_video_test_src_get_size),
	  (gst_video_test_src_smpte), (gst_video_test_src_snow),
	  (gst_video_test_src_unicolor), (paint_setup_AYUV),
	  (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
	  (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
	  * gst/videotestsrc/videotestsrc.h:
	  Add support for AYUV and the various RGBA formats. Initialise
	  fields of paintinfo structs allocated on the stack.
	  * tests/check/elements/videotestsrc.c: (right_shift_colour),
	  (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
	  (GST_START_TEST), (videotestsrc_suite):
	  Add unit tests for videotestsrc's RGB output.

2006-09-01 16:12:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  (gst_video_test_src_pattern_get_type),
	  (gst_video_test_src_set_pattern):
	  * gst/videotestsrc/gstvideotestsrc.h:
	  * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
	  (gst_video_test_src_black), (gst_video_test_src_white),
	  (gst_video_test_src_red), (gst_video_test_src_green),
	  (gst_video_test_src_blue):
	  * gst/videotestsrc/videotestsrc.h:
	  Add more uni-colour patterns ("white", "red", "green", and "blue").

2006-09-01 10:07:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
	  Original commit message from CVS:
	  * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
	  Fix stride for YVYU, should be word-aligned (#353658).

2006-08-31 14:37:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/adder/gstadder.c: Fix build.
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_src_event):
	  Fix build.

2006-08-31 12:39:17 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (forward_event_func),
	  (gst_adder_src_event), (gst_adder_collected),
	  (gst_adder_change_state):
	  * gst/adder/gstadder.h:
	  Remember the start position asked in the incoming seeks, so we can
	  output GST_EVENT_NEW_SEGMENT with a correct position value (instead
	  of assuming it will always be 0).

2006-08-31 12:31:00 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
	  (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
	  (gst_ogg_demux_loop):
	  Send the GST_EVENT_NEW_SEGMENT from the streaming thread.

2006-08-30 17:22:27 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_get_unit_size):
	  Return FALSE instead of returning a random false unit
	  size when the format isn't known/supported (even if
	  this shouldn't happen under normal circumstances).

2006-08-29 15:23:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
	  Original commit message from CVS:
	  Patch by: Tim-Philipp Müller  <tim at centricular dot net>
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
	  (gst_gnome_vfs_src_start):
	  Try harder to get the size from a uri by using _info_uri() when
	  _info_from_handle() does not give us enough info.
	  Also follow symlinks when getting the size.
	  Partially Fixes #332864.

2006-08-29 11:50:51 +0000  Viktor Peters <viktor.peters@gmail.com>

	  ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
	  Original commit message from CVS:
	  Patch by: Viktor Peters  <viktor dot peters at gmail dot com>
	  * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
	  (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
	  (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
	  (gst_alsa_mixer_set_record):
	  * ext/alsa/gstalsamixertrack.c:
	  (gst_alsa_mixer_track_update_alsa_capabilities),
	  (alsa_track_has_cap), (gst_alsa_mixer_track_new),
	  (gst_alsa_mixer_track_update):
	  * ext/alsa/gstalsamixertrack.h:
	  Improve and fix mixer track handling, in particular better handling
	  of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
	  track objects for tracks that have both capture and playback volume
	  (and label them differently as well so they're not mistakenly
	  assumed to be duplicates); classify mixer tracks that only affect
	  the audible volume of something (rather than the capture volume)
	  as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
	  for capture tracks to correspond to alsa-pswitch alsa-cswitch
	  (following the meaning documented in the mixer interface header
	  file); add support for alsa's exclusive cswitch groups; update/sync
	  state/flags better if mixer settings are changed by another
	  application. Fixes #336075.

2006-08-29 10:58:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c:
	  Improve docs: add section about BUFFERING messages sent by playbin.

2006-08-29 10:51:12 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
	  (gst_vorbis_enc_buffer_check_discontinuous),
	  (gst_vorbis_enc_chain):
	  Ignore explicit DISCONT marked on buffers (which is often spurious,
	  particularly when using multiple segments), in favour of solely
	  using the timestamps/durations.

2006-08-29 10:32:34 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
	  Don't rely on incoming buffers offset anymore, since it is completely
	  broken when using multiple segments.
	  Instead convert the incoming buffers timestamp to running time, and
	  then convert that value to the offsets.
	  Also inform GstSegment of the last outputted stop position, which is
	  needed if we received several segments with an unknown stop value.

2006-08-29 08:03:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
	  fix buffer unreffing on a header push failure

2006-08-28 16:17:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
	  (gst_audio_rate_chain):
	  Make the metadata of the buffer writable before changing its
	  flags.

2006-08-28 16:09:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	* ChangeLog:
	  Fix changelog with bugzilla bug it fixed.
	  Original commit message from CVS:
	  Fix changelog with bugzilla bug it fixed.

2006-08-28 16:08:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audiorate/gstaudiorate.c: Fix audiorate some more.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
	  (gst_audio_rate_setcaps), (gst_audio_rate_init),
	  (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
	  (gst_audio_rate_chain), (gst_audio_rate_change_state):
	  Fix audiorate some more.
	  Reset and resync counters on flush and READY.
	  Handle the DISCONT flag correctly.
	  Use GstSegment to track position.
	  Fail when not negotiated.

2006-08-25 16:48:28 +0000  Michael Smith <msmith@xiph.org>

	  gst/tcp/gstmultifdsink.c: Fix spelling.
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	  Fix spelling.
	  Remove accidently included debug line.

2006-08-25 16:39:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/tcp/gstmultifdsink.c: Small cleanups.
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
	  Small cleanups.
	  If a buffer is received with no caps, make the buffer metadata
	  writable and set the caps, making sure that we don't screw up the
	  refcounts.

2006-08-25 16:19:55 +0000  Michael Smith <msmith@xiph.org>

	  gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
	  Original commit message from CVS:
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
	  (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
	  Fix memory leaks and misleading debug messages, add a couple of
	  comments.
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
	  (gst_multi_fd_sink_render):
	  Do not use gst_buffer_make_writable() in a basesink render method,
	  as it may incorrectly unref the buffer. Instead, use convoluted
	  dance to avoid copying the buffer except when we need to.

2006-08-25 09:54:56 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c:
	  (gst_vorbis_enc_buffer_check_discontinuous):
	  Allow very small discontinuities in the timestamps. These we can't
	  do anything useful with anyway (because vorbis's timestamps have
	  only sample granularity), and are commonly produced by elements with
	  minor bugs. Allow up to 1/2 a sample out.
	  Fixes #351742.

2006-08-24 11:18:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
	  (play_scrub_toggle_cb), (main):
	  Add a checkbox to enable play scrubbing. Makes it possible to disable
	  normal scrubbing.

2006-08-23 19:37:50 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/.cvsignore: make buildbot happy
	  Original commit message from CVS:
	  * tests/check/elements/.cvsignore:
	  make buildbot happy

2006-08-23 16:43:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
	  Original commit message from CVS:
	  * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
	  (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
	  (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
	  (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
	  (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
	  (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
	  (gst_ogm_text_parse_strip_trailing_zeroes),
	  (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
	  (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
	  Refactor ogm parse, do better input checking, misc. clean-ups.
	  Cache incoming events and push them once the source pad has
	  been created. Don't pass unterminated strings to sscanf().
	  Strip trailing zeroes from subtitle text output, since they
	  are not valid UTF-8. Don't push vorbiscomment packets on
	  the subtitle text pad. Output perfect streams if possible.

2006-08-23 15:27:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
	  Original commit message from CVS:
	  * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
	  Waits for tasks to settle down so that we clean up correctly for
	  valgrind.

2006-08-23 15:11:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
	  Original commit message from CVS:
	  * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
	  Unit test fixes: \377 is more likely to fit into 8 bits than \777;
	  actually return return value in taglists_are_equal.

2006-08-23 12:14:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
	  Original commit message from CVS:
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Fix crash due to broken bitstream parsing on x86-64: can't make
	  any assumptions about sizeof(struct) due to alignment/packing
	  differences on different architectures. Fixes #351790.

2006-08-22 16:31:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
	  (gst_riff_parse_chunk), (gst_riff_parse_file_header),
	  (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
	  (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
	  (gst_riff_parse_info):
	  Protect public functions against bad input.
	  Do some cleanups.
	  Fix documentation.

2006-08-22 15:50:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-ids.h:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Add voxware audio IDs (even if we can't play it) (#351795).

2006-08-22 15:11:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c:
	  (gst_riff_create_video_template_caps),
	  (gst_riff_create_audio_template_caps),
	  (gst_riff_create_iavs_template_caps):
	  Const-ify some arrays and use G_N_ELEMENTS instead
	  of wasting oodles of RAM on terminator bits.

2006-08-22 08:27:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  (gst_tag_list_to_vorbiscomment_buffer):
	  * tests/check/libs/tag.c: (GST_START_TEST):
	  And the same for _to_vorbiscomment_buffer(): allow
	  id_data_len == 0 for speex.

2006-08-21 19:04:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/gdp/README:
	  adding a README
	  Original commit message from CVS:
	  adding a README

2006-08-21 19:01:41 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  Move GDP plugin to -base from -bad.  Closes #347783.
	  Original commit message from CVS:
	  * configure.ac:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * docs/plugins/inspect/plugin-gdp.xml:
	  * gst/gdp/Makefile.am:
	  * tests/check/Makefile.am:
	  Move GDP plugin to -base from -bad.  Closes #347783.

2006-08-21 18:34:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  (gst_tag_list_from_vorbiscomment_buffer):
	  Allow id_data_len == 0 (needed for vorbis comments in Speex files).
	  Also add some checks to make sure we don't memcmp() beyond the end of
	  vorbiscomment buffer if the ID to check for is larger than the buffer.
	  * tests/check/libs/tag.c: (GST_START_TEST):
	  Some more tests for gst_tag_list_from_vorbiscomment_buffer().

2006-08-21 16:39:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
	  (gst_vorbis_enc_set_metadata):
	  Use vorbis comment utility functions from libgsttag
	  instead of re-inventing the wheel (partially fixes #347091).

2006-08-21 11:42:12 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (GST_START_TEST):
	  Fix leaks. Wait for state transitions that might happen ASYNC, as well
	  as some that won't.

2006-08-21 10:32:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * docs/libs/gst-plugins-base-libs.types:
	  Don't try to GObject scan the netbuffer as it's not a GObject.
	  Fixes #351308.
	  * gst-libs/gst/netbuffer/gstnetbuffer.c:
	  * gst-libs/gst/netbuffer/gstnetbuffer.h:
	  Document GstNetBuffer.

2006-08-21 08:54:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (GST_START_TEST),
	  (audioconvert_suite):
	  Add testcase for caps-size-explosion

2006-08-20 13:05:44 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c:
	  (gst_audio_convert_get_unit_size), (set_structure_widths):
	  Lower debug, use g_assert in _get_unit_size
	  * gst/audioresample/gstaudioresample.c:
	  (audioresample_get_unit_size):
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_get_unit_size):
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
	  use g_assert in _get_unit_size

2006-08-18 21:21:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: fix bug number
	  Original commit message from CVS:
	  ChangeLog surgery: fix bug number

2006-08-18 16:43:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Document GstRTPBuffer.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
	  (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
	  (gst_rtp_buffer_get_payload_buffer):
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Document GstRTPBuffer.
	  Added function to efficiently strip payload headers.
	  API: gst_rtp_buffer_get_payload_subbuffer()

2006-08-17 16:52:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
	  (gst_tag_to_vorbis_comments):
	  Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
	  tags and deserialise them properly as well (#351768).
	  Add some more gtk-doc blurbs and also some g_return_if_fail().
	  * tests/check/libs/tag.c: (GST_START_TEST),
	  (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
	  More tests.

2006-08-17 15:43:40 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
	  Original commit message from CVS:
	  * ext/ogg/Makefile.am:
	  * ext/ogg/gstogg.c: (plugin_init):
	  * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
	  (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
	  (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
	  (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
	  (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
	  (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
	  Added ogg-in-avi parser element. Fixes #140139.
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
	  Fixed a bug in oggdemux debug code.
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	  (gst_riff_create_audio_template_caps):
	  Recognise Ogg in the AVI extensible wave format.

2006-08-17 10:00:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
	  Make buffer durations add up (duration should be next_ts-ts for
	  perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
	  from CVS.
	  * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
	  (test_buffer_timestamps), (cddabasesrc_suite):
	  Add unit test for the above.
	  * tests/check/Makefile.am:
	  Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
	  to see what happens.

2006-08-16 11:38:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/: Avoid setting and using a NULL device name.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
	  (gst_alsasink_open):
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
	  (gst_alsasrc_open):
	  Avoid setting and using a NULL device name.
	  Print more info when we fail to open a device.

2006-08-16 11:28:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  API: add gst_tag_parse_extended_comment() (#351426).
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/tag/tag.h:
	  * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
	  API: add gst_tag_parse_extended_comment() (#351426).
	  * tests/check/Makefile.am:
	  * tests/check/libs/.cvsignore:
	  * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
	  Add unit test for gst_tag_parse_extended_comment().

2006-08-15 19:20:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/: Fix leak (#351502).
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
	  Fix leak (#351502).

2006-08-15 17:21:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Document playbin.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * docs/plugins/gst-plugins-base-plugins.args:
	  * gst/playback/gstplaybin.c:
	  Document playbin.
	  * docs/plugins/inspect/plugin-adder.xml:
	  * docs/plugins/inspect/plugin-alsa.xml:
	  * docs/plugins/inspect/plugin-audioconvert.xml:
	  * docs/plugins/inspect/plugin-audiorate.xml:
	  * docs/plugins/inspect/plugin-audioresample.xml:
	  * docs/plugins/inspect/plugin-audiotestsrc.xml:
	  * docs/plugins/inspect/plugin-cdparanoia.xml:
	  * docs/plugins/inspect/plugin-decodebin.xml:
	  * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	  * docs/plugins/inspect/plugin-gnomevfs.xml:
	  * docs/plugins/inspect/plugin-ogg.xml:
	  * docs/plugins/inspect/plugin-pango.xml:
	  * docs/plugins/inspect/plugin-playbin.xml:
	  * docs/plugins/inspect/plugin-subparse.xml:
	  * docs/plugins/inspect/plugin-tcp.xml:
	  * docs/plugins/inspect/plugin-theora.xml:
	  * docs/plugins/inspect/plugin-typefindfunctions.xml:
	  * docs/plugins/inspect/plugin-video4linux.xml:
	  * docs/plugins/inspect/plugin-videorate.xml:
	  * docs/plugins/inspect/plugin-videoscale.xml:
	  * docs/plugins/inspect/plugin-videotestsrc.xml:
	  * docs/plugins/inspect/plugin-volume.xml:
	  * docs/plugins/inspect/plugin-vorbis.xml:
	  * docs/plugins/inspect/plugin-ximagesink.xml:
	  * docs/plugins/inspect/plugin-xvimagesink.xml:
	  Update to CVS version.

2006-08-14 17:54:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
	  (gst_play_bin_set_property), (gst_play_bin_get_property),
	  (value_list_append_structure_list),
	  (gst_play_bin_handle_redirect_message),
	  (gst_play_bin_handle_message):
	  Add "connection-speed" property; re-order redirect messages with
	  multiple redirect locations depending on the minimum bitrate if
	  that information is available and a connection speed is set
	  (#350399).

2006-08-14 11:41:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c:
	  Update max volume to the same value that the volume element uses.

2006-08-14 10:50:15 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsamixer.c: Less uglyness..
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
	  Less uglyness..

2006-08-14 10:49:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Add some more debug info.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
	  (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
	  (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
	  Add some more debug info.
	  Don't crash when a seek failed.
	  Actually return the result of the seek instead of TRUE.
	  Ignore multiple BOS pages with the same serial so that we don't create
	  the same stream multiple times.
	  Post an error when we fail to do the initial seek.

2006-08-13 14:34:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsa.c: Small code cleanup.
	  Original commit message from CVS:
	  * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
	  (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
	  Small code cleanup.
	  * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
	  (gst_alsa_mixer_new):
	  Remove hack that always set the device to hw:0*.
	  Properly find the card name for whatever device was configured.
	  Do some better debugging.
	  Fixes #350784.
	  * ext/alsa/gstalsamixerelement.c:
	  (gst_alsa_mixer_element_set_property),
	  (gst_alsa_mixer_element_change_state):
	  Cleanups.
	  Handle setting of a NULL device name better.

2006-08-11 15:53:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
	  Original commit message from CVS:
	  * gst/adder/gstadder.c:
	  Don't clip float values. Fixes #350900.

2006-08-11 15:33:17 +0000  Andy Wingo <wingo@pobox.com>

	  gst/tcp/gsttcp.c: Really fix the build?
	  Original commit message from CVS:
	  2006-08-11  Andy Wingo  <wingo@pobox.com>
	  * gst/tcp/gsttcp.c: Really fix the build?

2006-08-11 15:29:56 +0000  Andy Wingo <wingo@pobox.com>

	  gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
	  Original commit message from CVS:
	  2006-08-11  Andy Wingo  <wingo@pobox.com>
	  * gst/tcp/gsttcp.h: For now, always disable deprecation here --
	  fixes the build.

2006-08-10 13:01:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
	  Float caps shouldn't have a "signed" field.

2006-08-10 08:56:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
	  Implement SEEKING query in its most basic form, so that we can
	  at least check if we're seekable or not (#350655).

2006-08-09 14:42:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
	  The checks here are not even close to anything that would
	  justify MAXIMUM probability, lowering to POSSIBLE until someone
	  fixes the checks (case at hand: quicktime redirection files
	  might start with 00 00 01 XX and pass the checks here just
	  fine, see #350399).

2006-08-08 13:57:29 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
	  Original commit message from CVS:
	  * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
	  I forgot to include the file containing the #define :)
	  Now includes "config.h"

2006-08-08 13:45:44 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
	  Original commit message from CVS:
	  * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
	  Ignore test known to fail on PPC64. See #348114.

2006-08-08 08:41:13 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons  <sjoerd at luon net>
	  * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
	  Better detection for multipart/x-mixed-replace: accept leading
	  whitespaces before the boundary marker as well (as our very own
	  multipartmux used to produce) (#349068).

2006-08-07 08:26:03 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst-libs/gst/riff/: Detect DTS audio streams (#350157).
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha  <ganadist at chollian net>
	  * gst-libs/gst/riff/riff-ids.h:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
	  (gst_riff_create_audio_template_caps):
	  Detect DTS audio streams (#350157).

2006-08-05 17:08:05 +0000  Andy Wingo <wingo@pobox.com>

	  ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
	  Original commit message from CVS:
	  2006-08-05  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/gsttheoraparse.h:
	  * ext/theora/theoraparse.c (gst_theora_parse_class_init)
	  (theora_parse_dispose, theora_parse_set_property)
	  (theora_parse_get_property, theora_parse_munge_granulepos)
	  (theora_parse_push_buffer, theora_parse_change_state): Add a
	  property 'synchronization-points' to fix badly synchronized oggs.

2006-08-04 13:20:23 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay.  Fixes #349916.
	  Original commit message from CVS:
	  2006-08-04  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
	  Fix event parsing by gdpdepay.  Fixes #349916.

2006-08-03 15:04:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add a few tests for the channel position stuff in libgstaudio.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/libs/.cvsignore:
	  * tests/check/libs/audio.c: (structure_contains_channel_positions),
	  (fixed_caps_have_channel_positions), (GST_START_TEST),
	  (audio_suite), (main):
	  Add a few tests for the channel position stuff in libgstaudio.

2006-08-03 14:16:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
	  Original commit message from CVS:
	  * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
	  (gst_alsa_detect_channels):
	  * ext/alsa/gstalsasink.c:
	  Add support for cards that (only) do more than 8 channels,
	  like the Delta 44 (#345188).
	  * gst-libs/gst/audio/multichannel.c:
	  (gst_audio_check_channel_positions):
	  * gst-libs/gst/audio/multichannel.h:
	  API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
	  unspecified channel position and cannot be combined with any
	  of the other audio channel positions; adjust position layout
	  checks accordingly (#345188).

2006-08-03 11:15:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Recognise ancient RealAudio files (see #349779).

2006-08-03 09:01:25 +0000  Jens Granseuer <jensgr@gmx.net>

	  gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
	  Original commit message from CVS:
	  Patch by: Jens Granseuer  <jensgr at gmx net>
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Add typefinder for Interplay's MVE format (#348973).

2006-08-02 17:03:29 +0000  Marcel Moreaux <marcelm@luon.net>

	  gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
	  Original commit message from CVS:
	  Patch by: Marcel Moreaux <marcelm at luon dot net>
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_add_to_queue):
	  * gst-libs/gst/rtp/gstbasertpdepayload.h:
	  Handle RTP sequence number rollover.
	  Disable jitterbuffer by default.

2006-08-02 16:56:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/gdp/gstgdpdepay.c: Disable seeking.
	  Original commit message from CVS:
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
	  (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
	  (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
	  (gst_gdp_depay_change_state):
	  Disable seeking.
	  Small cleanups.
	  Clear adapter on disconts.
	  Clear caps when going to READY instead of NULL
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
	  (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
	  (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
	  (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
	  (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
	  (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
	  (gst_gdp_pay_change_state):
	  * gst/gdp/gstgdppay.h:
	  Reset payloader when going to READY.
	  Fix leaked buffers in ->queue on push errors.
	  Disable seeking.
	  Code cleanups.
	  Create packetizer in _init, free in _finalize.

2006-07-31 08:48:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
	  Original commit message from CVS:
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
	  (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
	  Consume all events except EOS because we generate events from
	  the gdp payload instead. Fixes #349204

2006-07-28 17:17:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c: (audioresample_stop),
	  (audioresample_set_caps):
	  Don't leak references to the incoming caps. Clean them up when
	  stopping.
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
	  (gst_video_scale_finalize):
	  Don't leak our temporary pixel buffer.
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
	  (GST_START_TEST), (simple_launch_lines_suite):
	  Fix leaks and re-enable the test for valgrind checking.

2006-07-28 16:39:31 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons  <sjoerd at luon net>
	  * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
	  (plugin_init):
	  Add typefind function for multipart/x-mixed-replace (#348916).

2006-07-28 14:14:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.c: Fix leak in duration query.
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_setcaps),
	  (gst_adder_query_duration):
	  Fix leak in duration query.
	  Reflow some docs and notes.

2006-07-28 13:42:48 +0000  Michael Smith <msmith@xiph.org>

	  tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
	  Original commit message from CVS:
	  * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
	  (vorbisenc_suite):
	  Enable Andy's extra vorbisenc test, now that it passes. Also fix one
	  aspect of it.

2006-07-28 12:48:21 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
	  (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
	  (gst_vorbis_enc_push_buffer),
	  (gst_vorbis_enc_buffer_check_discontinuous),
	  (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
	  * ext/vorbis/vorbisenc.h:
	  Handle discontinuities in the input vorbis stream correctly,
	  so that the output is properly timestamped (and has good granulepos
	  values). Needs some oggmux fixes too.

2006-07-27 10:52:52 +0000  Kai Vehmanen <kv2004@eca.cx>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
	  Original commit message from CVS:
	  patch by: Kai Vehmanen <kv2004 eca cx>
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_chain),
	  (gst_base_rtp_depayload_handle_sink_event),
	  (gst_base_rtp_depayload_change_state):
	  Don't send multiple newsegments with different formats.
	  Fixes #348677.

2006-07-26 15:20:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	  (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
	  Make seeking in ogg more accurate again by doing the more correct
	  granuletime to stream time conversion.

2006-07-26 10:59:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
	  (gst_multi_fd_sink_new_client):
	  debug a little more understandably
	  do not use goto as a substitute for break, especially if
	  break is also being used

2006-07-26 10:55:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gsttcp.c:
	  move a recurring normal event to LOG, where it should be
	  Original commit message from CVS:
	  move a recurring normal event to LOG, where it should be

2006-07-26 10:54:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/vorbis/vorbisdec.c:
	  tweak debug output
	  Original commit message from CVS:
	  tweak debug output

2006-07-26 10:52:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
	  Original commit message from CVS:
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
	  proxying get/set caps is the wrong thing to do, since we really
	  do change caps quite fundamentally
	  * tests/check/elements/gdpdepay.c:
	  * tests/check/elements/gdppay.c:
	  remove declaration of buffers, it's already done in gstcheck.h

2006-07-26 10:31:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Remove GLib-2.6 compatibility cruft.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
	  * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
	  Remove GLib-2.6 compatibility cruft.

2006-07-24 16:47:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  Don't try to align a sample to an unknown value.

2006-07-24 15:14:17 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
	  When the audio clock is slaved to another clock, never try to align
	  samples but trust the rate interpolation algorithm.

2006-07-24 14:34:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	  Don't try to calculate silence samples, base class does this much
	  better now.
	  * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	  (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
	  (gst_ring_buffer_acquire):
	  Calculate silence samples correctly.
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Add _CAST macro.

2006-07-22 17:01:12 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
	  Limit search for the first markup tag to the first few kB of
	  the file. If we don't find one there, it's highly unlikely that
	  this is an XML(-ish) file.

2006-07-21 17:04:06 +0000  Andy Wingo <wingo@pobox.com>

	  tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
	  Original commit message from CVS:
	  2006-07-21  Andy Wingo  <wingo@pobox.com>
	  * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
	  test to the one in vorbisenc. Also commented out.

2006-07-21 16:54:19 +0000  Andy Wingo <wingo@pobox.com>

	  tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
	  Original commit message from CVS:
	  2006-07-21  Andy Wingo  <wingo@pobox.com>
	  * tests/check/pipelines/vorbisenc.c:
	  (test_discontinuity): New test, commented out until Mike lands
	  some elite vorbisenc patches.

2006-07-21 15:59:24 +0000  Andy Wingo <wingo@pobox.com>

	  tests/check/pipelines/: Port to bufferstraw.
	  Original commit message from CVS:
	  2006-07-21  Andy Wingo  <wingo@pobox.com>
	  * tests/check/pipelines/vorbisenc.c:
	  * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
	  Bufferstraw was actually factored out of these tests. Now we share
	  code yay.

2006-07-21 11:03:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/theoradec.c: Better clipping.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (clip_buffer):
	  Better clipping.

2006-07-21 10:43:54 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstaudiosink.c: Fix leak.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
	  (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
	  (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
	  Fix leak.
	  Avoid type casting when we can.
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
	  Fix mem leak.

2006-07-20 16:57:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixerelement.c:
	  (gst_alsa_mixer_element_change_state):
	  Make state change fail if the specified device can't be opened
	  for some reason.

2006-07-20 10:42:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/test.c: Example of a small audio/video player using decodebin.
	  Original commit message from CVS:
	  * gst/playback/test.c: (gen_video_element), (gen_audio_element),
	  (cb_newpad), (main):
	  Example of a small audio/video player using decodebin.

2006-07-20 05:56:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-ids.h:
	  Add 'fact' chunk id

2006-07-19 18:20:43 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_chain),
	  (gst_base_rtp_depayload_change_state):
	  Don't assert when not negotiated but post a meaningfull
	  error message. Fixes #347918.
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  Add comment about better default MTU size.
	  * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
	  Small cleanups, start docs.

2006-07-19 14:46:36 +0000  Martin Szulecki <compiz@sukimashita.com>

	  sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
	  Original commit message from CVS:
	  Patch by: Martin Szulecki
	  * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
	  If "device-name" is requested and the device is not
	  open, try to temporarily open it to obtain this
	  information (#342494).

2006-07-19 12:25:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c:
	  Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
	  * gst-libs/gst/tag/gsttageditingprivate.h:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  Some more random const-ifications.

2006-07-18 19:48:48 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-ids.h:
	  * gst-libs/gst/riff/riff-media.c:
	  (gst_riff_create_video_template_caps):
	  Add more FOURCCs (sort list to make stuff easier to find),
	  add comment what those 16 bytes in struct _gst_riff_strh according to
	  one avi-dumper are

2006-07-17 14:17:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
	  Original commit message from CVS:
	  2006-07-17  Thomas Vander Stichele  <thomas at apestaart dot org>
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
	  remove parent_class setting, BOILERPLATE does this
	  (gst_gdp_pay_reset_streamheader):
	  fix typo in comment

2006-07-17 13:48:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/multichannel.c:
	  (gst_audio_check_channel_positions),
	  (gst_audio_fixate_channel_positions):
	  Const-ify two arrays.

2006-07-17 12:33:42 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
	  Original commit message from CVS:
	  * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
	  Fix typo, so that alsasink also advertises 8 channels
	  if that's supported (tags: can, worms, open, alsa, ph34r).

2006-07-17 12:01:04 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
	  (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
	  *sigh*, when is the compiler going to warn when the comments
	  are out-of-sync with the code.. Refix case of busted theora
	  headers with 0 granule pos.

2006-07-14 17:56:59 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_wait),
	  (gst_base_rtp_depayload_change_state),
	  (gst_base_rtp_depayload_set_property),
	  (gst_base_rtp_depayload_get_property):
	  Fix 99% cpu load by waiting for absolute times on the
	  clock. Fixes #347300.

2006-07-14 17:07:08 +0000  Andy Wingo <wingo@pobox.com>

	  ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
	  Original commit message from CVS:
	  2006-07-14  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/gsttheoraparse.h:
	  * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
	  (theora_parse_push_headers, theora_parse_clear_queue)
	  (theora_parse_drain_queue_prematurely, )
	  (theora_parse_sink_event, theora_parse_change_state): Queue events
	  until we initialized our state, like in vorbisparse.

2006-07-14 16:45:17 +0000  Iain * <iaingnome@gmail.com>

	  ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
	  Original commit message from CVS:
	  2006-07-14  Andy Wingo  <wingo@pobox.com>
	  * ext/vorbis/vorbisparse.h:
	  * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
	  (vorbis_parse_push_headers, vorbis_parse_clear_queue)
	  (vorbis_parse_drain_queue_prematurely, )
	  (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
	  until we have initialized our state. Fixes seeking after an
	  initial pad block.
	  2006-07-14  Andy Wingo  <wingo@pobox.com>
	  Patch by: Iain * <iaingnome@gmail.com>
	  * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.

2006-07-14 15:52:39 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Bump nano back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Bump nano back to CVS

=== release 0.10.9 ===

2006-07-14 15:51:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: releasing 0.10.9, "I walk the line"
	  Original commit message from CVS:
	  2006-07-13  Jan Schmidt <thaytan@mad.scientist.com>
	  * configure.ac:
	  releasing 0.10.9, "I walk the line"

2006-07-14 14:12:40 +0000  Michael Smith <msmith@xiph.org>

	  tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
	  Original commit message from CVS:
	  * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
	  Move a g_cond_signal to earlier to avoid sometimes deadlocking
	  (commonly happens when running this test under valgrind) when trying
	  to remove the buffer probe.

2006-07-14 10:34:36 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/gdp/Makefile.am:
	  build as a plugin, not a lib
	  Original commit message from CVS:
	  build as a plugin, not a lib

2006-07-13 16:43:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
	  Fix missing g_unlock from the previous commit

2006-07-13 16:34:04 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	  (gst_ximagesink_change_state):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	  (gst_xvimagesink_change_state):
	  Implement a locking order to ensure we always take the object lock
	  before the x_lock and never vice-versa.

2006-07-13 15:25:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  docs/plugins/: add more plugins and elements to docs
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
	  add more plugins and elements to docs
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
	  fix segfaults due to wrong g_free
	  add example
	  * gst/gdp/gstgdppay.c:
	  add example

2006-07-13 14:38:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (find_compatibles):
	  Fix a caps leak when linking (#347304)
	  * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	  (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
	  (gst_ximagesink_change_state):
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
	  (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
	  (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
	  Don't leak shared memory resources. Use the object lock to protect
	  against the xcontext disappearing while returning a buffer from the
	  pipeline. (#347304)

2006-07-12 14:20:43 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
	  (vorbis_handle_comment_packet):
	  gst_tag_list_merge() returns a new object. Take that into account when
	  using it. This avoids memleak.
	  Revert previous commit which is not needed.

2006-07-12 13:30:20 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
	  Reset the decoder in finalize so that all fields get cleared.

2006-07-12 13:24:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_set_clock),
	  (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
	  Don't try to post an error message when setting the clock fails
	  as this can happen when adding an element to a bin which will then
	  deadlock. Fixes #347296.

2006-07-12 13:04:15 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
	  (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
	  (vorbis_handle_type_packet):
	  Post tag messages on the bus even if we're not initialized.
	  If we're not initialized, we still postpone the event pushing of tags.

2006-07-12 11:28:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Revert last two changes that broke the freeze.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	  * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	  (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	  Revert last two changes that broke the freeze.

2006-07-12 10:59:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
	  basesink calculates silence sample correctly for us.

2006-07-12 10:58:42 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
	  (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
	  Calculate correct silence samples so we don't fill our ringbuffer
	  with noise.

2006-07-12 10:37:18 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
	  (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
	  (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
	  * ext/vorbis/vorbisdec.h:
	  Delay sending events (newsegment, tags) until the decoder is properly
	  initialized.
	  Fixes #347295

2006-07-11 22:40:13 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-07-11 21:04:16 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (get_float_mc_caps),
	  (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
	  Patch from #347221 adding a test for audioconvert
	  channel remappings.

2006-07-11 12:03:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
	  Original commit message from CVS:
	  * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
	  (gst_ssa_parse_parse_line):
	  Don't include the terminating NUL in the buffer size,
	  it's only there for extra paranoia (would add random
	  '*' characters at the end of each subtitle since the
	  terminator itself is not valid UTF-8 technically).
	  Also fix indenting after boilerplate macro.

2006-07-10 14:59:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (close_pad_link):
	  Also emit 'unknown-type' signal (which should really be
	  called unhandled-type) if we found potential decoders/demuxers
	  in the registry but none of them worked in the end (as in the
	  case where the plugins don't exist any longer but are still
	  listed in the registry). Fixes #329798.

2006-07-08 13:48:58 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/theora/theoraparse.c:
	  theoraparse.c (theora_parse_push_buffer)
	  Original commit message from CVS:
	  2006-07-08  Andy Wingo  <wingo@pobox.com>
	  * theoraparse.c (theora_parse_push_buffer)
	  (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
	  Add some more debugging. Fix granulepos reconstruction in the face
	  of discontinuities.

2006-07-06 15:54:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_class_init),
	  (gst_base_audio_sink_provide_clock):
	  Use gobject_class instead of G_OBJECT_CLASS (klass)
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_class_init), (gst_base_audio_src_init),
	  (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
	  (gst_base_audio_src_get_time),
	  (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
	  (gst_base_audio_src_create_ringbuffer):
	  Fix latency and buffer-time constants and properties ala basesink.
	  Implement pull based scheduling. Fixes #346527.
	  Set default blocksize in GstBaseSrc to 0, we default to pushing out
	  one segment.
	  Refuse slaving to another clock instead of silently not working.
	  Only provide a clock when we are actually able to do so.
	  Various small cleanups and compiler hints.

2006-07-06 13:23:07 +0000  Lutz Mueller <lutz@topfrose.de>

	  gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
	  Original commit message from CVS:
	  Patch by: Lutz Mueller <lutz at topfrose de>
	  * gst/typefind/gsttypefindfunctions.c: (html_type_find),
	  (plugin_init):
	  Add typefinding for text/html (#346581).

2006-07-06 13:12:02 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
	  (xml_check_first_element), (xml_type_find), (smil_type_find):
	  Fix SMIL typefinding, make xml_check_first_element() more
	  useful.

2006-07-06 13:04:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
	  (gst_play_base_bin_finalize), (decodebin_element_added_cb),
	  (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
	  * gst/playback/gstplaybasebin.h:
	  Protect list of elements with a subtitle-encoding property and
	  the subtitle encoding member itself with a lock of their own
	  instead of using the object lock. This prevents a dead-lock in
	  the element-remove callback in some circumstances when shutting
	  down playbin.

2006-07-05 20:11:13 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/common/libgsttag.def: Export some new functions.
	  Original commit message from CVS:
	  * win32/common/libgsttag.def:
	  Export some new functions.
	  * win32/vs6/libgstogg.dsp:
	  Add a link to libgsttag-0.10.lib.

2006-07-04 16:50:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsamixertrack.c: Some const-ification.
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
	  Some const-ification.

2006-07-04 14:06:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
	  Improve checking if we are dealing with a stream. Added some
	  more uris that need buffering.

2006-07-03 10:43:31 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/vorbis/vorbisdec.c: Remove unused variable.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
	  Remove unused variable.

2006-07-02 21:48:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  Makefile.am: include lcov.mak
	  Original commit message from CVS:
	  * Makefile.am:
	  include lcov.mak
	  * configure.ac:
	  add GCOV_LIBS to GST_LIBS

2006-07-02 11:08:58 +0000  Michael Sheldon <webmaster@mikeasoft.com>

	  ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
	  Original commit message from CVS:
	  Patch by: Michael Sheldon  <webmaster at mikeasoft com>
	  * ext/alsa/gstalsasrc.c:
	  Add 32 bps to template caps and increase channels range
	  from [1,2] to [1,MAX]. See #346326.

2006-06-30 12:04:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Recognise 'WMVA' video codec fourcc (#345879).

2006-06-29 12:21:06 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  Fixed nasty memory leak

2006-06-26 13:19:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/tcp/gsttcp.c: fix logging
	  Original commit message from CVS:
	  * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
	  (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
	  fix logging

2006-06-23 16:45:50 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
	  (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
	  (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
	  Protect remove_fakesink using a mutex, so that we don't try and
	  remove the fakesink simultaneously from multiple threads.
	  When going from READY to PAUSED, restore the fakesink, so that
	  it is there when decodebin gets reused.

2006-06-23 09:53:09 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  * gst/tcp/gstmultifdsink.c:
	  * gst/tcp/gsttcpclientsink.c:
	  * gst/tcp/gsttcpclientsrc.c:
	  * gst/tcp/gsttcpserversink.c:
	  * gst/tcp/gsttcpserversrc.c:
	  * gst/videorate/gstvideorate.c:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  * sys/v4l/gstv4ljpegsrc.c:
	  * sys/v4l/gstv4lmjpegsink.c:
	  * sys/v4l/gstv4lsrc.c:
	  * tests/examples/seek/scrubby.c:
	  * tests/examples/seek/seek.c:
	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).

2006-06-23 09:09:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
	  Original commit message from CVS:
	  * ext/directfb/dfbvideosink.c:
	  * ext/gsm/gstgsmdec.c:
	  * ext/gsm/gstgsmenc.c:
	  * ext/libmms/gstmms.c:
	  * ext/neon/gstneonhttpsrc.c:
	  * ext/theora/theoradec.c:
	  * gst/freeze/gstfreeze.c:
	  * gst/gdp/gstgdpdepay.c:
	  * gst/gdp/gstgdppay.c:
	  * sys/glsink/glimagesink.c:
	  Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
	  and fix one GObject boilerplate macro.

2006-06-22 12:13:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
	  Second field in GEnumValue shouldn't be a description,
	  but a stringified version of the enum value.

2006-06-22 12:03:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	  (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
	  (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
	  Avoid type checking in buffer casts.
	  Avoid caps copy in buffer_alloc when we can.
	  Use pad_peer_accept.

2006-06-22 11:01:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/tag.h:
	  Oops, make that 'Since: 0.10.9'.

2006-06-22 10:55:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  API: add GstTagImageType enum to describe images contained in image tags (#345641).
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/tag/tag.h:
	  * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
	  (gst_tag_image_type_get_type):
	  API: add GstTagImageType enum to describe images contained
	  in image tags (#345641).

2006-06-22 10:31:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
	  Fix warnings with gst-inspect: "buffers-min" property
	  should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
	  typo in property description.

2006-06-22 10:10:51 +0000  Cody Russell <bratsche@gnome.org>

	  gst/: Avoid unnecessary class cast check in class_init functions (#337747).
	  Original commit message from CVS:
	  Patch by: Cody Russell <bratsche at gnome org>
	  * gst/audioresample/gstaudioresample.c:
	  (gst_audioresample_class_init):
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_class_init):
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
	  * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
	  * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
	  * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
	  * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
	  * gst/videotestsrc/gstvideotestsrc.c:
	  (gst_video_test_src_class_init):
	  * gst/volume/gstvolume.c: (gst_volume_class_init):
	  Avoid unnecessary class cast check in class_init
	  functions (#337747).

2006-06-21 18:39:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
	  (gst_text_overlay_video_chain):
	  g_markup_escape_text() REALLY doesn't like non-UTF8 input
	  and doesn't validate its input either (and neither did
	  textoverlay it seems). Let's do that then and fix #345206.

2006-06-19 17:12:57 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
	  (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
	  (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
	  (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
	  (find_syncframe), (find_limits), (assign_value),
	  (count_burst_unit), (gst_multi_fd_sink_new_client),
	  (gst_multi_fd_sink_handle_client_write),
	  (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
	  (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
	  (gst_multi_fd_sink_change_state):
	  * gst/tcp/gstmultifdsink.h:
	  Added shiny new burst-on-connect methods.
	  Add properties to control the minimal amount of data queued.
	  Small cleanups.
	  API: bytes-min property
	  API: time-min property
	  API: buffers-min property
	  API: burst-unit property
	  API: burst-value property
	  API: add-full signal
	  * gst/tcp/gsttcp-marshal.list:
	  Added new marshaller code for the new signal.
	  * tests/check/elements/multifdsink.c: (GST_START_TEST),
	  (multifdsink_suite):
	  Added testcases for new burst methods.

2006-06-19 11:35:47 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  update for latest changes
	  Original commit message from CVS:
	  update for latest changes

2006-06-19 09:57:50 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/theora/theoradec.c: Implement clipping for accurate seeking.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
	  Implement clipping for accurate seeking.
	  Closes #345225

2006-06-19 09:08:05 +0000  Philip Jaegenstedt <philip@lysator.liu.se>

	  gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
	  Original commit message from CVS:
	  Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
	  (gst_video_scale_transform):
	  Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131

2006-06-17 14:18:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery
	  Original commit message from CVS:
	  ChangeLog surgery

2006-06-17 14:13:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
	  Original commit message from CVS:
	  * configure.ac:
	  Fix --disable-extern (can't set conditionals conditionally,
	  #343602).

2006-06-16 15:43:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
	  Original commit message from CVS:
	  * tests/check/elements/audioresample.c: (test_reuse),
	  (audioresample_suite):
	  Add test case for bug #342789 fixed below.

2006-06-16 15:17:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c:
	  (gst_audioresample_class_init), (gst_audioresample_init),
	  (audioresample_start), (audioresample_stop),
	  (gst_audioresample_set_property), (gst_audioresample_get_property):
	  Implement GstBaseTransform::start and ::stop so that audioresample
	  can clear its internal state properly and be reused insted of
	  causing non-negotiated errors with playbin under some circumstances
	  (#342789).
	  * tests/check/elements/audioresample.c: (setup_audioresample),
	  (cleanup_audioresample):
	  Need to set element state here so that ::start and ::stop are
	  called.

2006-06-16 13:59:29 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha <ganadist at chollian dot net>
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
	  Parse extra data better, apparently it's right behind
	  the normal strf header size. Fixes #343500.

2006-06-16 11:04:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (set_hwparams):
	  If we fail to set the buffer_time and period_time alsa
	  parameters, post a warning and leave alsa select a
	  default instead of failing. Fixes #342085

2006-06-16 10:30:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
	  Original commit message from CVS:
	  ChangeLog surgery (it wouldn't have crashed, just shown bogus values)

2006-06-16 10:20:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/cdda/gstcddabasesrc.h:
	  Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
	  out in the header file and shouldn't be listed in the docs.
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	  Fix it so that it doesn't crash in the debug statement.

2006-06-16 10:02:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/: add remaining symbols into correct setions
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * docs/libs/gst-plugins-base-libs.types:
	  add remaining symbols into correct setions
	  * gst-libs/gst/audio/gstringbuffer.c:
	  fix incomplete docs
	  * gst-libs/gst/audio/gstringbuffer.h:
	  comment out not yet implemented function
	  * gst-libs/gst/floatcast/floatcast.h:
	  * gst-libs/gst/netbuffer/gstnetbuffer.c:
	  add short descriptions
	  * gst-libs/gst/interfaces/propertyprobe.c:
	  fix return value docs
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	  simplify debug logging
	  * gst-libs/gst/riff/riff-read.h:
	  sync function prototype and docs
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  remove left over symbol

2006-06-16 09:56:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Use GST_PLUGIN_DOCS macro in configure.ac, add
	  Original commit message from CVS:
	  * autogen.sh:
	  * configure.ac:
	  * docs/Makefile.am:
	  Use GST_PLUGIN_DOCS macro in configure.ac, add
	  --enable-plugin-docs default to autogen.sh and use
	  ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).

2006-06-15 15:27:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
	  (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
	  (gst_ogg_demux_loop):
	  Combine GstFlowReturn from the source pads to give a
	  meaningfull result to the upstream peer or to stop the
	  processing task in case of errors.

2006-06-14 14:49:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
	  Original commit message from CVS:
	  * gst/playback/gststreaminfo.c: (cb_probe):
	  Try GST_TAG_CODEC as fallback when extracting the
	  codec name; more debug info.

2006-06-14 14:34:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
	  Original commit message from CVS:
	  * ext/ogg/Makefile.am:
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Extract language tags from ogm subtitle streams, so that
	  the subtitle menu choices are labelled correctly in
	  Totem (fixes #344708).

2006-06-14 09:13:54 +0000  Alessandro Decina <alessandro@nnva.org>

	  ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
	  Original commit message from CVS:
	  Patch by: Alessandro Decina <alessandro at nnva dot org>
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
	  (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
	  (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
	  (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
	  Fix various leaks. Fixes #343699.
	  Add x-smoke mime type.

2006-06-14 08:17:45 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-ids.h:
	  Add IDs for 'bext' chunks (see #343837).

2006-06-12 12:44:38 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha  <ganadist at chollian net>
	  * gst/subparse/samiparse.c: (sami_context_pop_state),
	  (handle_start_font), (end_sami_element):
	  Honour font face tags in SAMI subtitles (#344503).

2006-06-11 20:41:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  po/POTFILES.in: add missing files containing translatable strings
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  add missing files containing translatable strings

2006-06-11 19:55:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
	  Original commit message from CVS:
	  * docs/libs/tmpl/.cvsignore:
	  we don't want those *.sgml files in CVS either

2006-06-11 19:44:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ignore more
	  Original commit message from CVS:
	  * docs/libs/.cvsignore:
	  * tests/check/elements/.cvsignore:
	  * tests/check/libs/.cvsignore:
	  ignore more

2006-06-11 18:33:27 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  also commiting the changed Makefile.am (added more libs to the
	  doc-build)

2006-06-11 17:08:26 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  docs/libs/: first batch of reordering things, add index & hierarchy
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * docs/libs/gst-plugins-base-libs.types:
	  first batch of reordering things, add index & hierarchy

2006-06-11 14:08:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* ext/alsa/Makefile.am:
	* ext/cdparanoia/Makefile.am:
	* ext/gnomevfs/Makefile.am:
	* ext/libvisual/Makefile.am:
	* ext/ogg/Makefile.am:
	* ext/pango/Makefile.am:
	* ext/theora/Makefile.am:
	* ext/vorbis/Makefile.am:
	* sys/v4l/Makefile.am:
	* sys/ximage/Makefile.am:
	* sys/xvimage/Makefile.am:
	  further clean up build
	  Original commit message from CVS:
	  further clean up build

2006-06-11 12:14:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
	  Original commit message from CVS:
	  * configure.ac:
	  use GST_PKG_CHECK_MODULES, cleans up output

2006-06-11 12:10:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* win32/common/config.h:
	  update to cvs
	  Original commit message from CVS:
	  update to cvs

2006-06-10 18:52:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
	  Add support for burn:// URIs (#343385); const-ify things a bit,
	  use G_N_ELEMENTS instead of hard-coded array size.

2006-06-10 18:25:07 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha  <ganadist at chollian net>
	  * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
	  Fix up broken entities before passing them to libxml *sigh*.
	  (#343303).

2006-06-09 18:52:35 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	  back to trunk
	  Original commit message from CVS:
	  back to trunk

=== release 0.10.8 ===

2006-06-09 18:49:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* win32/common/config.h:
	  releasing 0.10.8
	  Original commit message from CVS:
	  releasing 0.10.8

2006-06-07 11:04:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  0.10.7.2 prerelease
	  Original commit message from CVS:
	  * configure.ac:
	  * po/af.po:
	  * po/az.po:
	  * po/cs.po:
	  * po/en_GB.po:
	  * po/hu.po:
	  * po/it.po:
	  * po/nb.po:
	  * po/nl.po:
	  * po/or.po:
	  * po/sq.po:
	  * po/sr.po:
	  * po/sv.po:
	  * po/uk.po:
	  * po/vi.po:
	  * win32/common/config.h:
	  0.10.7.2 prerelease

2006-06-07 11:03:03 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  move last template doc snippets to source code and delete them
	  Original commit message from CVS:
	  * docs/libs/tmpl/gstaudio.sgml:
	  * docs/libs/tmpl/gstcolorbalance.sgml:
	  * docs/libs/tmpl/gstmixer.sgml:
	  * docs/libs/tmpl/gstringbuffer.sgml:
	  * docs/libs/tmpl/gsttuner.sgml:
	  * docs/libs/tmpl/gstxoverlay.sgml:
	  * gst-libs/gst/audio/audio.c:
	  * gst-libs/gst/audio/gstringbuffer.c:
	  * gst-libs/gst/interfaces/colorbalance.c:
	  * gst-libs/gst/interfaces/mixer.c:
	  * gst-libs/gst/interfaces/tuner.c:
	  * gst-libs/gst/interfaces/xoverlay.c:
	  move last template doc snippets to source code and delete them

2006-06-06 16:26:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/gdp/gstgdppay.c:
	  adapt to new api
	  Original commit message from CVS:
	  adapt to new api

2006-06-06 14:39:15 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: enable building of GDP elements
	  Original commit message from CVS:
	  * configure.ac:
	  enable building of GDP elements
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
	  (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
	  (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
	  (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
	  (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
	  (gst_gdp_pay_change_state):
	  * gst/gdp/gstgdppay.h:
	  add version 1.0

2006-06-06 11:13:18 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
	  Original commit message from CVS:
	  * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
	  (theora_parse_drain_queue):
	  Mark DELTA_UNIT on non-keyframes.

2006-06-03 21:06:49 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
	  * gst-libs/gst/audio/gstbaseaudiosink.h:
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
	  (gst_ring_buffer_samples_done):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Document better the fact that latency_time and buffer_time are values
	  stored in microseconds, and not the usual GStreamer nanoseconds.
	  Change the variables (compatibly) that store them from GstClockTime
	  to guint64 to make it more clear that they're not storing clock times.
	  Also, remove the bogus property description that says the user can
	  specify -1 to get the default value, since that's never been the case.
	  When computing the default segment size for the ring buffer, make it
	  an integer number of samples.
	  When the sub-class indicates a delay greater than the number of
	  samples we've written return 0 from the audio sink get_time method.

2006-06-02 17:01:02 +0000  Michael Smith <msmith@xiph.org>

	  tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (set_channel_positions),
	  (get_float_mc_caps), (get_int_mc_caps):
	  * tests/check/elements/audioresample.c:
	  * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
	  * tests/check/elements/videorate.c:
	  * tests/check/elements/videotestsrc.c: (GST_START_TEST):
	  * tests/check/elements/volume.c:
	  * tests/check/elements/vorbisdec.c:
	  * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
	  Don't busy-wait in tests; this was causing test timeouts very
	  frequently when running under valgrind.

2006-06-02 16:45:59 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/gdp/gstgdpdepay.c:
	* gst/gdp/gstgdppay.h:
	  small fixes
	  Original commit message from CVS:
	  small fixes

2006-06-02 16:35:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  fail_if_can_read is racy
	  Original commit message from CVS:
	  fail_if_can_read is racy

2006-06-02 16:26:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/tcp/: make multifdsink properly deal with streamheader:
	  Original commit message from CVS:
	  * gst/tcp/README:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
	  (gst_multi_fd_sink_remove_client_link),
	  (gst_multi_fd_sink_client_queue_caps),
	  (gst_multi_fd_sink_client_queue_buffer),
	  (gst_multi_fd_sink_handle_client_write),
	  (gst_multi_fd_sink_render):
	  * gst/tcp/gstmultifdsink.h:
	  make multifdsink properly deal with streamheader:
	  - streamheader is taken from caps
	  - buffers marked with IN_CAPS are not sent
	  - streamheaders are sent, on connection, from the caps of the
	  buffer where the client gets positioned to
	  - further streamheader changes are done every time the client
	  will receive a buffer with different caps
	  * tests/check/elements/multifdsink.c: (GST_START_TEST),
	  (gst_multifdsink_create_streamheader):
	  add tests for this

2006-06-02 15:06:59 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
	  Reinstate limit on channel count. Vorbis does not define the meaning
	  of > 6 channels, so they're just independent channels. Gstreamer
	  currently has no mechanism to represent N independent channels.

2006-06-02 14:23:34 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
	  Don't arbitrarily restrict channel counts and rate in vorbis.
	  In terms of effects likely on real-world files, this fixes 96kHz
	  playback of vorbis.

2006-06-02 14:19:18 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioconvert/audioconvert.c: More correct float->int conversion.
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c: (float):
	  More correct float->int conversion.

2006-06-02 14:07:42 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
	  Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
	  value. Fixes g-critical on trying to play back ogg containing
	  unknown codec.

2006-06-02 10:34:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
	  (setup_source):
	  * gst/playback/gstplaybasebin.h:
	  Make the subtitle detection work from any thread so we don't
	  deadlock. Fixes #343397.

2006-06-02 10:28:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
	  Original commit message from CVS:
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
	  (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
	  (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
	  (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
	  (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
	  (gst_gdp_pay_get_property):
	  add crc-header and crc-payload properties
	  don't error out on some things that are recoverable
	  * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
	  add test for crc

2006-06-02 09:17:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gsttcp.c:
	  show type number when packet is of the wrong type
	  Original commit message from CVS:
	  show type number when packet is of the wrong type

2006-06-01 23:04:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right.  Even a drunk can do it ! Add LIBOI...
	  Original commit message from CVS:
	  * gst/volume/Makefile.am:
	  Seriously, it's not *that* hard to get compilation right.  Even
	  a drunk can do it ! Add LIBOIL CFLAGS and LIBS

2006-06-01 22:00:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
	  Original commit message from CVS:
	  * ext/alsaspdif/alsaspdifsink.h:
	  * ext/amrwb/gstamrwbdec.h:
	  * ext/amrwb/gstamrwbenc.h:
	  * ext/amrwb/gstamrwbparse.h:
	  * ext/arts/gst_arts.h:
	  * ext/artsd/gstartsdsink.h:
	  * ext/audiofile/gstafparse.h:
	  * ext/audiofile/gstafsink.h:
	  * ext/audiofile/gstafsrc.h:
	  * ext/audioresample/gstaudioresample.h:
	  * ext/bz2/gstbz2dec.h:
	  * ext/bz2/gstbz2enc.h:
	  * ext/dirac/gstdiracdec.h:
	  * ext/directfb/dfbvideosink.h:
	  * ext/divx/gstdivxdec.h:
	  * ext/divx/gstdivxenc.h:
	  * ext/dts/gstdtsdec.h:
	  * ext/faac/gstfaac.h:
	  * ext/gsm/gstgsmdec.h:
	  * ext/gsm/gstgsmenc.h:
	  * ext/ivorbis/vorbisenc.h:
	  * ext/libfame/gstlibfame.h:
	  * ext/nas/nassink.h:
	  * ext/neon/gstneonhttpsrc.h:
	  * ext/polyp/polypsink.h:
	  * ext/sdl/sdlaudiosink.h:
	  * ext/sdl/sdlvideosink.h:
	  * ext/shout/gstshout.h:
	  * ext/snapshot/gstsnapshot.h:
	  * ext/sndfile/gstsf.h:
	  * ext/swfdec/gstswfdec.h:
	  * ext/tarkin/gsttarkindec.h:
	  * ext/tarkin/gsttarkinenc.h:
	  * ext/theora/theoradec.h:
	  * ext/wavpack/gstwavpackdec.h:
	  * ext/wavpack/gstwavpackparse.h:
	  * ext/xine/gstxine.h:
	  * ext/xvid/gstxviddec.h:
	  * ext/xvid/gstxvidenc.h:
	  * gst/cdxaparse/gstcdxaparse.h:
	  * gst/cdxaparse/gstcdxastrip.h:
	  * gst/colorspace/gstcolorspace.h:
	  * gst/festival/gstfestival.h:
	  * gst/freeze/gstfreeze.h:
	  * gst/gdp/gstgdpdepay.h:
	  * gst/gdp/gstgdppay.h:
	  * gst/modplug/gstmodplug.h:
	  * gst/mpeg1sys/gstmpeg1systemencode.h:
	  * gst/mpeg1videoparse/gstmp1videoparse.h:
	  * gst/mpeg2sub/gstmpeg2subt.h:
	  * gst/mpegaudioparse/gstmpegaudioparse.h:
	  * gst/multifilesink/gstmultifilesink.h:
	  * gst/overlay/gstoverlay.h:
	  * gst/playondemand/gstplayondemand.h:
	  * gst/qtdemux/qtdemux.h:
	  * gst/rtjpeg/gstrtjpegdec.h:
	  * gst/rtjpeg/gstrtjpegenc.h:
	  * gst/smooth/gstsmooth.h:
	  * gst/smoothwave/gstsmoothwave.h:
	  * gst/spectrum/gstspectrum.h:
	  * gst/speed/gstspeed.h:
	  * gst/stereo/gststereo.h:
	  * gst/switch/gstswitch.h:
	  * gst/tta/gstttadec.h:
	  * gst/tta/gstttaparse.h:
	  * gst/videodrop/gstvideodrop.h:
	  * gst/xingheader/gstxingmux.h:
	  * sys/directdraw/gstdirectdrawsink.h:
	  * sys/directsound/gstdirectsoundsink.h:
	  * sys/dxr3/dxr3audiosink.h:
	  * sys/dxr3/dxr3spusink.h:
	  * sys/dxr3/dxr3videosink.h:
	  * sys/qcam/gstqcamsrc.h:
	  * sys/vcd/vcdsrc.h:
	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass

2006-06-01 20:39:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c: (volume_choose_func),
	  (volume_update_real_volume), (gst_volume_class_init),
	  (gst_volume_init), (volume_process_float), (volume_process_int16),
	  (volume_process_int16_clamp), (volume_set_caps),
	  (volume_transform_ip), (plugin_init):
	  * gst/volume/gstvolume.h:
	  rewrite the passthrough check, split _int16 and _int16_clamp, fix
	  another property desc., remove unused param from process function
	  * tests/check/elements/volume.c: (volume_suite):
	  reactivate the passthrough test

2006-06-01 19:19:51 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixerelement.h:
	  * ext/alsa/gstalsamixeroptions.h:
	  * ext/alsa/gstalsamixertrack.h:
	  * ext/gnomevfs/gstgnomevfssink.h:
	  * ext/gnomevfs/gstgnomevfssrc.h:
	  * ext/theora/gsttheoradec.h:
	  * ext/theora/gsttheoraenc.h:
	  * ext/theora/gsttheoraparse.h:
	  * ext/vorbis/vorbisparse.h:
	  * gst-libs/gst/audio/gstaudioclock.h:
	  * gst-libs/gst/audio/gstaudiofilter.h:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  * gst/audioconvert/gstaudioconvert.h:
	  * gst/audioresample/gstaudioresample.h:
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
	  * gst/playback/gststreamselector.h:
	  * gst/tcp/gstmultifdsink.h:
	  * gst/tcp/gsttcpclientsink.h:
	  * gst/tcp/gsttcpclientsrc.h:
	  * gst/tcp/gsttcpserversink.h:
	  * gst/tcp/gsttcpserversrc.h:
	  * gst/videorate/gstvideorate.h:
	  * gst/videoscale/gstvideoscale.h:
	  * gst/videotestsrc/gstvideotestsrc.h:
	  * gst/volume/gstvolume.h:
	  * sys/v4l/gstv4ljpegsrc.h:
	  * sys/v4l/gstv4lmjpegsink.h:
	  * sys/v4l/gstv4lmjpegsrc.h:
	  * sys/v4l/gstv4lsrc.h:
	  * sys/ximage/ximagesink.h:
	  * sys/xvimage/xvimagesink.h:
	  * tests/old/testsuite/alsa/sinesrc.h:
	  Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass

2006-05-31 16:56:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  remove wrong commit
	  Original commit message from CVS:
	  remove wrong commit

2006-05-31 16:21:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libvisual/visual.c: Handle DISCONT.
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_reset),
	  (gst_visual_sink_setcaps), (gst_visual_sink_event),
	  (gst_visual_src_event), (get_buffer), (gst_visual_chain):
	  Handle DISCONT.
	  Use running time before doing QoS.
	  Handle mono too.

2006-05-31 14:17:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  set a magic variable to indicate we know the docs are incomplete

2006-05-30 20:33:59 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
	  Original commit message from CVS:
	  * win32/common/libgstvideo.def:
	  export gst_video_calculate_display_ratio
	  * win32/vs6/libgstvideoscale.dsp:
	  add link to libgstvideo-0.10.lib

2006-05-30 19:00:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gen_source_element):
	  Throw a more comprehensible error for rtsp:// URIs (rather
	  than erroring out with a negotiation error later on) until
	  we fix playbin to handle rtspsrc etc.

2006-05-30 16:09:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/pango/gsttextoverlay.c: Added some FIXMEs.
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
	  (gst_text_overlay_text_event):
	  Added some FIXMEs.

2006-05-30 16:07:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.*: Implement release_request_pad.
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
	  (gst_adder_request_new_pad), (gst_adder_release_pad):
	  * gst/adder/gstadder.h:
	  Implement release_request_pad.
	  Make padcounter atomic.
	  * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
	  Added check for release_pad in adder.

2006-05-30 16:04:14 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Fix build again.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
	  Fix build again.

2006-05-30 14:59:24 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
	  (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
	  (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	  (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
	  (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
	  (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
	  (gst_ogg_demux_bisect_forward_serialno),
	  (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
	  (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
	  add more debugging
	  clean up printf formats for granulepos and serialno

2006-05-30 14:31:43 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	* tests/check/generic/states.c:
	  properly fail if we can't make an element
	  Original commit message from CVS:
	  properly fail if we can't make an element

2006-05-30 13:22:58 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (raw_caps_factory),
	  (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
	  (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
	  (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
	  (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
	  * ext/vorbis/vorbisenc.h:
	  Multi-channel caps negotiation, so we can do proper multichannel
	  vorbis encoding, negotiated through audioconvert.

2006-05-30 11:45:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
	  Original commit message from CVS:
	  * tests/check/elements/adder.c: (test_event_message_received),
	  (test_play_twice_message_received), (GST_START_TEST),
	  (adder_suite):
	  Added check to show that #339935 is fixed with ongoing
	  adder and collectpads fixes.

2006-05-29 17:19:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.c: Don't leak pad name.
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_request_new_pad):
	  Don't leak pad name.

2006-05-29 15:49:53 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.c: Fix adder seeking.
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_query_duration),
	  (forward_event_func), (forward_event), (gst_adder_src_event):
	  Fix adder seeking.
	  Make query/seeking code threadsafe.
	  * tests/check/Makefile.am:
	  * tests/check/elements/adder.c: (test_event_message_received),
	  (GST_START_TEST), (test_play_twice_message_received):
	  Fix adder test case.

2006-05-29 13:21:00 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha  <ganadist at chollian net>
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	  (gst_play_base_bin_init), (gst_play_base_bin_dispose),
	  (set_encoding_element), (decodebin_element_added_cb),
	  (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
	  (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
	  * gst/playback/gstplaybasebin.h:
	  Add 'subtitle-encoding' property to playbin, so applications can
	  force a subtitle encoding for non-UTF8 subtitles (#342268).
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
	  (gst_sub_parse_set_property):
	  Rename recently-added 'encoding' property to 'subtitle-encoding'
	  (so it can be proxied by playbin/decodebin in a generic way
	  with less danger of false positives).

2006-05-29 11:04:48 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
	  Original commit message from CVS:
	  * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
	  (append_with_other_format), (set_structure_widths),
	  (gst_audio_convert_transform_caps):
	  Patch from #341562: give more specific audio caps in get_caps, so
	  that basetransform  can make better decisions on what caps to
	  negotiate.

2006-05-28 20:04:12 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/volume.c: make it compile again
	  Original commit message from CVS:
	  * tests/check/elements/volume.c:
	  make it compile again

2006-05-28 19:56:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/volume.c: disable test until #343196 gets resolved
	  Original commit message from CVS:
	  * tests/check/elements/volume.c: (volume_suite):
	  disable test until #343196 gets resolved

2006-05-28 19:42:27 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/adder/gstadder.c: Make it easier to copy&paste
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_get_type):
	  Make it easier to copy&paste
	  * gst/volume/Makefile.am:
	  * gst/volume/gstvolume.c: (volume_update_real_volume),
	  (gst_volume_set_volume), (gst_volume_set_mute),
	  (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
	  (volume_transform_ip), (volume_update_mute),
	  (volume_update_volume):
	  * gst/volume/gstvolume.h:
	  Add own debug category, move duplicate code to helper function, fix
	  property texts, add more comments and prepare ffor liboil-goodness
	  * tests/check/Makefile.am:
	  * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
	  add test for mute and passtrough case, be a bit more verbose to track
	  failure
	  * tests/check/generic/states.c: (GST_START_TEST):
	  catch elements that fail to instantiate

2006-05-28 09:37:18 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
	  Original commit message from CVS:
	  * tests/check/pipelines/simple-launch-lines.c:
	  * tests/check/pipelines/theoraenc.c:
	  * tests/check/pipelines/vorbisenc.c:
	  Comment out tests using parse_launch() if core was built without
	  parsing capabilities.

2006-05-27 13:34:03 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Extra bonus points for whoever explains to ensonic that you are meant
	  to test unit tests thoroughly before commiting them, especially if
	  you know it's going to break.
	  De-activated element/adder tests.

2006-05-27 13:09:16 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
	  Marking caps conversion issues as GST_WARNING is way too verbose,
	  Moving them to GST_LOG.

2006-05-27 11:26:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
	  Original commit message from CVS:
	  * README:
	  Replace current README (containing the release notes from
	  some 0.9.x version) with a proper README taken from the core.

2006-05-26 15:52:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.c: Small cleanups.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	  (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
	  (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
	  (vorbis_dec_change_state):
	  Small cleanups.
	  Add some FIXMEs
	  Clip output samples to segment boundaries.

2006-05-26 11:17:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
	  (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
	  Improve the errors produced on bad output, including some human
	  readable description strings.
	  Handle the (theoretical for ximagesink) case where the XServer
	  has a different idea about the size required for a particular
	  frame and gives us too small a memory allocation.

2006-05-26 10:18:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Mention bugs fixed by previous commit
	  Original commit message from CVS:
	  Mention bugs fixed by previous commit

2006-05-26 09:40:35 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	  (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
	  (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
	  (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
	  Improve the errors produced on bad output, including some human
	  readable description strings.
	  Handle RGB Xv formats properly by transforming them into our
	  big-endian caps description.
	  Use gst_caps_truncate to ensure that we never try and choose a
	  non-fixed caps in buffer_alloc.
	  Handle the case where the XServer has a different idea about the size
	  required for a particular frame and gives us too small a memory
	  allocation.
	  Use -1 to indicate 'no image format', because 0 is a valid XServer
	  image format number.
	  Put RGB Xv formats at the end of the caps, so that we always prefer
	  YUV format frames.
	  Iterate the available Xv Encodings to determine the maximum width and
	  height, and then return that in our caps.

2006-05-25 16:50:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
	  When there is only one unfinished pad and it receives an event that
	  doesn't match our requirements, we need to set alldone=FALSE so that
	  the fakesink is not removed yet.

2006-05-25 09:32:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
	  Use gst_type_find_helper_for_buffer() to find the type
	  of stream from the first packet.
	  * configure.ac:
	  Bump requirements to core CVS (needed for vorbis
	  typefinding to work).

2006-05-24 08:34:53 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
	  Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
	  Else they play perfectly fine with qtdemux.

2006-05-23 20:38:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  make more debug catagories static
	  Original commit message from CVS:
	  * ext/theora/theoradec.c:
	  * ext/theora/theoraenc.c:
	  * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
	  * gst/audiorate/gstaudiorate.c:
	  make more debug catagories static
	  * tests/check/Makefile.am:
	  * tests/check/elements/adder.c: (message_received),
	  (test_event_message_received), (GST_START_TEST),
	  (test_play_twice_message_received), (adder_suite):
	  added test case for using element twice, extra bonus points for anyone
	  who can make these test run reliably

2006-05-23 15:18:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_dec_chain):
	  Make work with time-stamped input buffers that do not
	  have a granulepos in BUFFER_OFFSET_END (like theora
	  buffers coming from matroskademux). Fixes #342448.

2006-05-22 15:53:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
	  Original commit message from CVS:
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
	  (gst_gdp_depay_change_state):
	  * gst/gdp/gstgdpdepay.h:
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
	  (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
	  (gst_gdp_pay_change_state):
	  * gst/gdp/gstgdppay.h:
	  Handle error cases when calling functions
	  do downwards state change after parent's change_state
	  * tests/check/elements/gdpdepay.c: (GST_START_TEST):
	  * tests/check/elements/gdppay.c: (GST_START_TEST):
	  clean up more

2006-05-22 13:25:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  adding GDP payloader and depayloader.  Build integration will follow later when the GDP issues for core are sorted out.
	  Original commit message from CVS:
	  * gst/gdp/Makefile.am:
	  * gst/gdp/gstgdp.c: (plugin_init):
	  * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
	  (gst_gdp_depay_class_init), (gst_gdp_depay_init),
	  (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
	  (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
	  * gst/gdp/gstgdpdepay.h:
	  * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
	  (gst_gdp_pay_class_init), (gst_gdp_pay_init),
	  (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
	  (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
	  (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
	  (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
	  (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
	  (gst_gdp_pay_plugin_init):
	  * gst/gdp/gstgdppay.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
	  (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
	  (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
	  * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
	  (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
	  (main):
	  adding GDP payloader and depayloader.  Build integration will
	  follow later when the GDP issues for core are sorted out.

2006-05-22 11:42:03 +0000  Peter Kjellerstedt <pkj@axis.com>

	  gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
	  Original commit message from CVS:
	  Patch by: Peter Kjellerstedt  <pkj at axis com>
	  * gst/tcp/Makefile.am:
	  fdstresstest doesn't need Gtk+, fix compilation if
	  gtk is not available (#342566).

2006-05-19 17:57:56 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  80 line columns
	  Removed redundant floor()

2006-05-19 15:00:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	  On second thought, just skip JUNK chunks automatically, so
	  the caller doesn't have to handle this. Fixes #342345.
	  Also, return GST_FLOW_UNEXPECTED if we get a short read,
	  not GST_FLOW_ERROR.

2006-05-19 13:37:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-read.c: Don't bail out on JUNK chunks with a size of 0 (would try to pull_range 0 bytes before...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
	  Don't bail out on JUNK chunks with a size of 0 (would try to
	  pull_range 0 bytes before, which sources don't like too much).
	  See #342345.

2006-05-19 13:02:46 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
	  Original commit message from CVS:
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
	  Use the gstutil scaling function to preserve 64 bits while calculating
	  output width and height from the display-aspect-ratio. (A continuation
	  of #341542)

2006-05-19 11:50:17 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
	  Original commit message from CVS:
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
	  (gst_xvimagesink_buffer_alloc):
	  * sys/xvimage/xvimagesink.h:
	  When performing buffer allocations, remember the caps and image format
	  we return so that if the same caps are asked for next time we can
	  return them immediately without doing any caps intersections.

2006-05-18 23:00:02 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/README: Some new documentation
	  Original commit message from CVS:
	  2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
	  * gst-libs/gst/rtp/README:
	  Some new documentation
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  New RTP audio base payloader class. Supports frame or sample based codecs.
	  Not enabled in Makefile.am until approved.

2006-05-18 20:30:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
	  Original commit message from CVS:
	  * tests/check/elements/alsa.c: (test_device_property_probe):
	  Fix test case: don't try to free NULL GValueArray when there
	  are no devices.

2006-05-18 19:21:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/alsa.c: (test_device_property_probe),
	  (alsa_suite), (main):
	  Add simple test that runs a device property probe on alsasrc,
	  alsasink and alsamixer. Disable valgrind check for now (too
	  many leaks in libasound, and valgrind ignored my suppressions
	  additions).

2006-05-18 17:19:39 +0000  Martin Szulecki <gnomebugzilla@sukimashita.com>

	  ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
	  Original commit message from CVS:
	  * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
	  (gst_alsa_device_property_probe_probe_property),
	  (gst_alsa_device_property_probe_needs_probe),
	  (gst_alsa_device_property_probe_get_values),
	  (gst_alsa_type_add_device_property_probe_interface):
	  * ext/alsa/gstalsadeviceprobe.h:
	  * ext/alsa/gstalsamixerelement.c:
	  (gst_alsa_mixer_element_init_interfaces):
	  * ext/alsa/gstalsamixerelement.h:
	  Clean up and simplify alsa device probing. Make it actually work
	  for multiple classes. Don't cache results any longer.
	  * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
	  (gst_alsasink_init):
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
	  (gst_alsasrc_interface_supported), (gst_implements_interface_init),
	  (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
	  Make alsasink and alsasrc implement the GstPropertyProbe interface
	  for device probing (#342181).
	  Patch by: Martin Szulecki  <gnomebugzilla at sukimashita com>

2006-05-18 10:05:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
	  Original commit message from CVS:
	  * gst/subparse/samiparse.c: (handle_start_font):
	  Don't ignore return value of strtol (++compiler_happiness).

2006-05-17 17:49:10 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha  <ganadist chollian net>
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	  (gst_sub_parse_class_init), (gst_sub_parse_init),
	  (gst_sub_parse_set_property), (gst_sub_parse_get_property),
	  (convert_encoding):
	  * gst/subparse/gstsubparse.h:
	  Add 'encoding' property (#341681).
	  * gst/subparse/samiparse.c: (characters_sami):
	  Output is pango markup, so we need to escape text
	  between tags (#342143).

2006-05-16 17:34:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/multichannel.c:
	  (gst_audio_check_channel_positions):
	  It's okay to have caps with channels=1 and a channel position
	  different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
	  (deinterleavers might want to keep the position in the caps,
	  so that they can be re-interleaved again properly later).
	  Leave check for unexpected 2-channel layouts intact for now.

2006-05-16 16:28:10 +0000  Zaheer Abbas Merali <zaheerabbas@merali.org>

	  gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
	  Original commit message from CVS:
	  2006-05-16  Zaheer Abbas Merali  <zaheerabbas at merali dot org>
	  * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
	  Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
	  basesrc can do its job correctly.

2006-05-16 15:52:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
	  Original commit message from CVS:
	  * ext/alsa/Makefile.am:
	  * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
	  (gst_alsa_detect_formats), (get_channel_free_structure),
	  (caps_add_channel_configuration), (gst_alsa_detect_channels),
	  (gst_alsa_probe_supported_formats):
	  * ext/alsa/gstalsa.h:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
	  Refactor and improve caps probing code: probe signedness
	  when we probe the supported formats/widths; set endianness
	  to the one we actually probed for (ie. cpu endianness).
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
	  (gst_alsasrc_close):
	  * ext/alsa/gstalsasrc.h:
	  Implement caps probing for alsasrc.

2006-05-15 17:42:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (gst_theora_dec_reset),
	  (theora_dec_src_query), (theora_dec_src_event),
	  (theora_dec_sink_event), (theora_handle_comment_packet),
	  (theora_handle_data_packet), (theora_dec_change_state):
	  Cleanups, add some G_LIKELY.
	  Use segment helpers instead of our own wrong code.
	  Clear queued buffers on seek and READY.
	  * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
	  (vorbis_dec_convert), (vorbis_dec_src_query),
	  (vorbis_dec_src_event), (vorbis_dec_sink_event),
	  (vorbis_handle_comment_packet), (vorbis_dec_push),
	  (vorbis_handle_data_packet), (vorbis_dec_chain),
	  (vorbis_dec_change_state):
	  * ext/vorbis/vorbisdec.h:
	  Remove old useless packetno variable.
	  Do position query properly.
	  Add some G_LIKELY.
	  Do cleanup of queued buffers in new helper function
	  and use it.

2006-05-15 17:17:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
	  Query supported sample rates. Fixes #341732.

2006-05-15 17:01:02 +0000  Julien Moutte <julien@moutte.net>

	  gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
	  Original commit message from CVS:
	  2006-05-15  Julien MOUTTE  <julien@moutte.net>
	  * gst/playback/gstdecodebin.c: (cleanup_decodebin),
	  (gst_decode_bin_change_state): Make decodebin reusable
	  when going from PAUSE_TO_READY and then back to PAUSED.
	  Fixes #331678.

2006-05-15 16:49:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
	  (vorbis_dec_convert), (vorbis_dec_src_query),
	  (vorbis_dec_sink_query), (vorbis_dec_src_event),
	  (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
	  (vorbis_dec_clean_queued), (vorbis_dec_push),
	  (vorbis_handle_data_packet), (vorbis_dec_change_state):
	  Cleanups. Use refcounting and DEBUG_OBJECT.
	  Reset segment on flush, use code methods instead of our
	  own wrong version.
	  Fix potential memleak.

2006-05-15 16:46:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
	  (gst_alsasink_init):
	  * ext/alsa/gstalsasink.h:
	  Don't leak allocated snd_output_t structure if there's
	  more than one alsasink instance at a time (#341873).
	  Also fix GObject macros in header file.

2006-05-15 15:31:30 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c:
	  (gst_sub_parse_data_format_autodetect):
	  Don't use libxml functions in the typefinding code.

2006-05-15 15:01:08 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
	  Fix seeking performance in the case where a non-header
	  packet has a 0 granulepos (busted theora case).
	  Fixes #341719

2006-05-15 14:19:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c:
	  (gst_sub_parse_data_format_autodetect):
	  Improve SAMI typefinding: handle case where there are
	  whitespaces or newlines in front of the first <SAMI>
	  tag (#169936).

2006-05-15 12:18:13 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
	  Original commit message from CVS:
	  * configure.ac:
	  Build video4linux plugin even if there's no XVIDEO, just
	  without implementing the GstXOverlay interface (#334002).

2006-05-15 10:17:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add tentative support for libvisual-0.4 (#336881).
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
	  (plugin_init):
	  Add tentative support for libvisual-0.4 (#336881).

2006-05-15 09:41:03 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha <ganadist at chollian net>
	  * gst/subparse/samiparse.c: (handle_start_font):
	  Need to map "silver" colour explicitly (#169936).

2006-05-15 09:14:35 +0000  Young-Ho Cha <ganadist@chollian.net>

	  gst/subparse/: Add support for SAMI subtitles (#169936).
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha  <ganadist at chollian net>
	  * gst/subparse/Makefile.am:
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	  (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
	  (gst_sub_parse_format_autodetect), (feed_textbuf),
	  (gst_subparse_type_find), (plugin_init):
	  * gst/subparse/gstsubparse.h:
	  * gst/subparse/samiparse.c:
	  * gst/subparse/samiparse.h:
	  Add support for SAMI subtitles (#169936).

2006-05-14 21:18:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* win32/common/config.h:
	  update config.h
	  Original commit message from CVS:
	  update config.h

2006-05-14 21:18:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/README:
	  fix mistakes in README
	  Original commit message from CVS:
	  fix mistakes in README

2006-05-14 18:15:17 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
	  Original commit message from CVS:
	  * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
	  Fix #341696: crash when mixing L+R+C to mono or stereo.
	  * tests/check/Makefile.am:
	  * tests/check/elements/audioconvert.c: (set_channel_positions),
	  (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
	  (audioconvert_suite):
	  Add test for the above, including some generic framework bits for
	  testing multichannel things.

2006-05-14 16:05:47 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  Back to CVS
	  Original commit message from CVS:
	  Back to CVS

=== release 0.10.7 ===

2006-05-14 16:00:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: releasing 0.10.7, "Leave the gun"
	  Original commit message from CVS:
	  2006-05-14  Jan Schmidt <thaytan@mad.scientist.com>
	  * configure.ac:
	  releasing 0.10.7, "Leave the gun"

2006-05-14 15:55:16 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* common:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-05-12 22:22:37 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fix the build.
	  Original commit message from CVS:
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
	  Fix the build.

2006-05-12 21:30:00 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
	  * gst-libs/gst/video/video.h:
	  * gst/videoscale/Makefile.am:
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
	  * tests/check/Makefile.am:
	  * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
	  (main):
	  Fix integer overflow problem with pixel-aspect-ratio calculations
	  in videoscale and xvimagesink (#341542)

2006-05-12 16:56:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c:
	  Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).

2006-05-12 10:39:08 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/MANIFEST: update win32 files listing
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  update win32 files listing

2006-05-11 21:47:01 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  disable failing check on gentoo64
	  Original commit message from CVS:
	  disable failing check on gentoo64

2006-05-11 21:35:44 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  disable failing check on gentoo64
	  Original commit message from CVS:
	  disable failing check on gentoo64

2006-05-11 21:20:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  macros show the correct line
	  Original commit message from CVS:
	  macros show the correct line

2006-05-11 21:04:08 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  macros show the correct line
	  Original commit message from CVS:
	  macros show the correct line

2006-05-11 21:01:05 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
	  Original commit message from CVS:
	  2006-05-11  Thomas Vander Stichele  <thomas at apestaart dot org>
	  patch by: Sjoerd Simons (sjoerd@luon.net)
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
	  (group_create), (group_destroy), (add_stream),
	  (gst_play_base_bin_get_property),
	  (gst_play_base_bin_get_streaminfo_value_array):
	  * gst/playback/gstplaybasebin.h:
	  API: GstPlayBaseBin::stream-info-value-array property
	  use a more bindings-friendly way of exposing streaminfo
	  using a GValueArray.  Tested in ipython.
	  Closes #341114

2006-05-11 19:44:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  fix some type warnings
	  Original commit message from CVS:
	  fix some type warnings

2006-05-11 19:38:22 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
	  (queue_underrun_cb), (queue_filled_cb):
	  Also catch queue underruns but don't do anything yet.
	  Refactor and comment queue enlarging code a bit.
	  * gst/playback/gstplaybasebin.c: (queue_overrun),
	  (queue_threshold_reached), (queue_out_of_data),
	  (gen_preroll_element):
	  If a queue over/underruns check that we don't create nasty
	  deadlocks when the min-threshold is not reached but the
	  max-bytes is. In those cases disable max-bytes when we
	  know that the queue is fed timed data.
	  Add more comments.

2006-05-11 18:06:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gen_audio_element):
	  Make playbin automatically plug an 'audioresample'
	  element before the audio sink as well. This solves
	  problems with sinks that only accept a very specific
	  sample rate, like esdsink (e.g. #340379).

2006-05-11 16:04:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gen_source_element):
	  Make http sources send special headers so that we receive
	  icecast metadata if the http stream is an icecast stream
	  (otherwise the server will just ignore them). This also
	  means that from now on users will need the 'icydemux'
	  element from gst-plugins-good installed if they want to
	  listen to icecast radio streams. (#341432, #333657).

2006-05-11 12:34:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gstmultifdsink.c:
	  more commenting
	  Original commit message from CVS:
	  more commenting

2006-05-11 11:40:39 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
	  (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
	  remove stupid example from docs - it should come with a simple
	  C program instead.
	  Clean up/fix docs
	  * tests/check/elements/multifdsink.c: (wait_bytes_served),
	  (fail_if_can_read), (GST_START_TEST),
	  (gst_multifdsink_create_streamheader), (multifdsink_suite):
	  add a test for changing streamheader which exposes a bug in
	  multifdsink

2006-05-11 10:33:46 +0000  Michael Smith <msmith@xiph.org>

	  ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
	  (gst_gnome_vfs_src_received_headers_callback):
	  * ext/gnomevfs/gstgnomevfssrc.h:
	  Don't set icy-caps unless we have a sane interval value. Move
	  interval to a local variable; we never use it outside this function.

2006-05-11 10:14:20 +0000  Wim Taymans <wim.taymans@gmail.com>

	  sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
	  Register special buffer types along with the objects so
	  that they are not registered at runtime from N different
	  streaming threads since they are not threadsafe.

2006-05-10 18:31:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/multifdsink.c:
	  set caps and plug leaks
	  Original commit message from CVS:
	  set caps and plug leaks

2006-05-10 18:16:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
	  Original commit message from CVS:
	  * tests/check/elements/multifdsink.c: (wait_bytes_served),
	  (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
	  add two more tests, one doing streamheader

2006-05-10 16:34:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
	  clean up the bufqueue when shutting down
	  * tests/check/Makefile.am:
	  * tests/check/elements/multifdsink.c: (setup_multifdsink),
	  (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
	  (main):
	  add a test for the leak that was just fixed

2006-05-10 15:16:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gstmultifdsink.c:
	  DEBUG_FUNCPTR'ing
	  Original commit message from CVS:
	  DEBUG_FUNCPTR'ing

2006-05-10 15:14:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gstmultifdsink.h:
	  whitespace fixes
	  Original commit message from CVS:
	  whitespace fixes

2006-05-10 11:54:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_setcaps),
	  (gst_adder_query_duration), (gst_adder_query), (forward_event),
	  (gst_adder_src_event), (gst_adder_sink_event),
	  (gst_adder_class_init), (gst_adder_finalize),
	  (gst_adder_request_new_pad), (gst_adder_collected):
	  * gst/adder/gstadder.h:
	  Updated some docs. Added comments and FIXMEs all over the place.
	  Improve debugging info.
	  Fix leak on finalize by not calling the parent.
	  Implement duration query.
	  Make event forwarding threadsafe.
	  Correctly send NEWSEGMENT at start and after flush.
	  Handle EOS correctly.
	  Post error when not negotiated.
	  * tests/check/elements/adder.c: (GST_START_TEST):
	  Added FIXME in the test.

2006-05-09 19:24:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Const-ify GEnumValue and GFlagsValue arrays. Use
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
	  (gst_text_overlay_halign_get_type),
	  (gst_text_overlay_wrap_mode_get_type):
	  * ext/theora/theoradec.c: (theora_handle_type_packet),
	  (theora_handle_data_packet):
	  * ext/theora/theoraenc.c: (gst_border_mode_get_type),
	  (theora_enc_sink_setcaps), (theora_enc_chain):
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_mode_get_type):
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  (gst_audiostestsrc_wave_get_type):
	  * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
	  * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
	  * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
	  (gst_sync_method_get_type), (gst_unit_type_get_type),
	  (gst_client_status_get_type):
	  * gst/videoscale/gstvideoscale.c:
	  (gst_video_scale_method_get_type):
	  * gst/videotestsrc/gstvideotestsrc.c:
	  (gst_video_test_src_pattern_get_type):
	  * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
	  (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
	  (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
	  (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
	  (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
	  (paint_setup_RGB565), (paint_setup_xRGB1555):
	  Const-ify GEnumValue and GFlagsValue arrays. Use
	  GST_ROUND_UP_* macros instead of home-made ones.

2006-05-09 17:40:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Require core CVS for the new newsegment stuff.
	  Original commit message from CVS:
	  * configure.ac:
	  Require core CVS for the new newsegment stuff.

2006-05-09 17:30:48 +0000  Sjoerd Simons <sjoerd@luon.net>

	  gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
	  Original commit message from CVS:
	  Patch by: Sjoerd Simons  <sjoerd at luon net>
	  * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
	  Register nick for enum value (#341160).

2006-05-09 16:46:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
	  (plugin_init):
	  backout typefind patch #340375
	  * tests/check/elements/adder.c: (message_received),
	  (GST_START_TEST), (adder_suite):
	  redo, signal-handling of test

2006-05-09 16:14:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_request_new_pad),
	  (gst_adder_collected):
	  * gst/adder/gstadder.h:
	  Remove bogus segment merging and forwarding, we don't
	  care about timestamps anyway and we just produce a
	  continuous stream.
	  Also create a nice NEWSEGMENT event when we start.
	  Use _scale_int some more.

2006-05-09 11:59:13 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
	  Original commit message from CVS:
	  * tests/icles/stress-xoverlay.c:
	  Fix if core was built without parsing support.

2006-05-09 11:37:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Add SEDG (Samsung MPEG-4) fourcc.

2006-05-09 11:31:47 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/examples/volume/volume.c: Fox if core was built without parsing support.
	  Original commit message from CVS:
	  * tests/examples/volume/volume.c:
	  Fox if core was built without parsing support.
	  * tests/examples/seek/seek.c:
	  Disable the parse_launch example if core was built without parsing
	  support.

2006-05-09 11:21:24 +0000  Edward Hervey <bilboed@bilboed.com>

	  tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c:
	  Disable the parse_launch example if core was built without parsing
	  support.

2006-05-08 15:51:15 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/libs/tmpl/gstcolorbalance.sgml:
	* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	* gst/tcp/gstmultifdsink.c:
	* gst/videoscale/gstvideoscale.c:
	  doc reparagraphing and DEBUG_FUNCPTRing
	  Original commit message from CVS:
	  doc reparagraphing and DEBUG_FUNCPTRing

2006-05-08 11:51:43 +0000  Edward Hervey <bilboed@bilboed.com>

	  autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
	  Original commit message from CVS:
	  * autogen.sh: (CONFIGURE_DEF_OPT):
	  libtoolize on Darwin/MacOSX is called glibtoolize

2006-05-07 17:39:04 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
	  Disable the adder test, until the build-slaves posses the kindness to
	  either like it or to give valid reason for not doing so

2006-05-07 17:25:56 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
	  Original commit message from CVS:
	  * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
	  (adder_suite):
	  Shuffle NULL state change around and raise timeout more

2006-05-07 17:07:03 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
	  (mp4_type_find), (plugin_init):
	  Add typefind to distinguish between "audio/x-m4a" and new type
	  "video/mp4". Fixes #340375
	  * tests/check/elements/adder.c: (adder_suite):
	  Raise timeout to make buildbot happy

2006-05-07 16:39:36 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_sink_event),
	  (gst_adder_request_new_pad), (gst_adder_change_state):
	  * gst/adder/gstadder.h:
	  * tests/check/Makefile.am:
	  * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
	  (adder_suite), (main):
	  Add sink-event handling to adder. It tries to merge incomming
	  newsegment-events. Added test to check if segment_done is comming
	  through.

2006-05-05 16:34:15 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/theora/theoraparse.c:
	* ext/vorbis/vorbisparse.c:
	  ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
	  Original commit message from CVS:
	  2006-05-05  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/theoraparse.c (gst_theora_parse_init)
	  (theora_parse_src_convert, theora_parse_src_query):
	  * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
	  (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
	  query functions on the source pads of the theora and vorbis parse
	  elements. Fixes position querying when doing a remux.

2006-05-05 13:46:37 +0000  Michael Smith <msmith@xiph.org>

	  ext/theora/theoraparse.c: Fix flushing.
	  Original commit message from CVS:
	  * ext/theora/theoraparse.c: (parse_granulepos),
	  (theora_parse_drain_queue_prematurely),
	  (theora_parse_queue_buffer), (theora_parse_sink_event):
	  Fix flushing.
	  Fix invalid granulepos outputs when starting with a non-keyframe.

2006-05-05 12:37:35 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
	  (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
	  Rearrange MPEG system stream detection, fixing some memleaks in the
	  process.
	  Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
	  they clean up their data correctly.
	  Remove unused ogganx caps and move the 'is_annodex' check to inside
	  the 'is_ogg' if statement.

2006-05-05 11:33:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (cleanup_decodebin):
	  Properly remove ghostpads. Fixes #340392

2006-05-04 18:43:58 +0000  David Schleef <ds@schleef.org>

	  gst/typefind/gsttypefindfunctions.c:
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c:

2006-05-03 16:32:19 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
	  (mpeg_ts_probe_headers), (mpeg_ts_type_find):
	  When typefinding an MP3 in push-based mode, don't penalise the
	  probability down to 74% when we found 5 valid frames just because we
	  can't peek the end of the file.
	  Make the probability for detecting MPEG Transport Streams based on the
	  number of sequential headers we successfully detected.

2006-05-03 15:52:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
	  (vorbis_dec_push), (vorbis_dec_chain):
	  Still produce an error when we receive an empty packet.

2006-05-03 15:34:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
	  (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
	  (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
	  Mark buffers with DISCONT after seek and after activating new
	  chains.
	  * ext/theora/gsttheoradec.h:
	  * ext/theora/theoradec.c: (gst_theora_dec_reset),
	  (theora_get_query_types), (theora_dec_sink_event),
	  (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
	  (theora_dec_change_state):
	  Fix frame counter.
	  Detect and mark DISCONT buffers.
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	  (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
	  (vorbis_dec_change_state):
	  * ext/vorbis/vorbisdec.h:
	  Use GstSegment.
	  Detect and mark DISCONT buffers.
	  Don't crash on 0 sized buffers.

2006-05-03 08:58:13 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
	  Original commit message from CVS:
	  * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
	  (volume_transform_ip):
	  Increase "volume" property to 10.0. Fixes #340369.
	  Set the process function to NULL when capsnego fails so that
	  we properly error out.

2006-05-02 18:15:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (add_sink):
	  * gst/playback/test.c: (main):
	  * gst/playback/test5.c: (dump_element_stats):
	  * gst/playback/test6.c: (main):
	  free cpas using gst_caps_unref, don't leak caps-strings

2006-05-02 06:33:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-libs/gst/rtp/gstbasertppayload.c:
	  some RTP debug
	  Original commit message from CVS:
	  some RTP debug

2006-05-01 19:08:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
	  (plugin_init):
	  Refine musepack typefinding a bit. Return MAXIMUM
	  probability when we detect stream version 7 to make
	  sure the mpeg audio typefinder doesn't trump us.

2006-04-29 16:25:58 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Protect against unexpected NULL strf_data buffer.

2006-04-29 13:09:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (verify_convert),
	  (GST_START_TEST):
	  interpret the out[] buffer in the order the bytes are actually
	  put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
	  Other tests should use BYTE_ORDER since the array is filled in
	  with actual values

2006-04-29 12:10:52 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/elements/audioconvert.c:
	  dump expected data when audioconvert test fails
	  Original commit message from CVS:
	  dump expected data when audioconvert test fails

2006-04-29 11:55:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (verify_convert),
	  (GST_START_TEST):
	  when a test fails, give an indication of which it is

2006-04-29 09:48:16 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/gstoggmux.c:
	* ext/theora/theoraenc.c:
	  add another include
	  Original commit message from CVS:
	  add another include

2006-04-29 01:24:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/subparse/gstssaparse.c:
	  atoi() needs stdlib.h
	  Original commit message from CVS:
	  atoi() needs stdlib.h

2006-04-29 01:18:05 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/playback/test4.c:
	* gst/playback/test5.c:
	* gst/playback/test6.c:
	  exit needs stdlib.h
	  Original commit message from CVS:
	  exit needs stdlib.h

2006-04-29 01:10:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  compile fix; strtol() needs <stdlib.h>

2006-04-29 01:04:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* common:
	* docs/Makefile.am:
	* docs/libs/Makefile.am:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/plugins/Makefile.am:
	* docs/upload.mak:
	  use common upload.mak
	  Original commit message from CVS:
	  use common upload.mak

2006-04-28 19:46:37 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  make GstElementDetails const
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixerelement.c:
	  * ext/alsa/gstalsasrc.c:
	  * ext/cdparanoia/gstcdparanoiasrc.c:
	  * ext/gnomevfs/gstgnomevfssink.c:
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  * ext/ogg/gstoggdemux.c:
	  * ext/ogg/gstoggmux.c:
	  * ext/ogg/gstoggparse.c:
	  * ext/ogg/gstogmparse.c:
	  * ext/pango/gstclockoverlay.c:
	  * ext/pango/gsttextoverlay.c:
	  * ext/pango/gsttextrender.c:
	  * ext/pango/gsttimeoverlay.c:
	  * ext/theora/theoradec.c:
	  * ext/theora/theoraenc.c:
	  * ext/vorbis/vorbisdec.c:
	  * ext/vorbis/vorbisenc.c:
	  * gst-libs/gst/audio/gstaudiofilter.c:
	  * gst-libs/gst/audio/gstaudiofiltertemplate.c:
	  * gst/audioconvert/gstaudioconvert.c:
	  * gst/audiorate/gstaudiorate.c:
	  * gst/audioresample/gstaudioresample.c:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  * gst/playback/gstdecodebin.c:
	  * gst/playback/gstplaybin.c:
	  * gst/playback/gststreamselector.c:
	  * gst/subparse/gstsubparse.c:
	  * gst/tcp/gstmultifdsink.c:
	  * gst/tcp/gsttcpclientsink.c:
	  * gst/tcp/gsttcpclientsrc.c:
	  * gst/tcp/gsttcpserversink.c:
	  * gst/tcp/gsttcpserversrc.c:
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  * gst/videorate/gstvideorate.c:
	  * gst/videoscale/gstvideoscale.c:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  * gst/volume/gstvolume.c:
	  * sys/v4l/gstv4ljpegsrc.c:
	  * sys/v4l/gstv4lmjpegsink.c:
	  * sys/v4l/gstv4lmjpegsrc.c:
	  * sys/v4l/gstv4lsrc.c:
	  * sys/ximage/ximagesink.c:
	  * sys/xvimage/xvimagesink.c:
	  * tests/check/libs/cddabasesrc.c:
	  make GstElementDetails const

2006-04-28 19:17:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
	  Original commit message from CVS:
	  * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
	  (gst_adder_init):
	  send events from src-pad to all sink-pads fixes #338657

2006-04-28 19:08:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
	  (alsasink_parse_spec):
	  query witdh capabilities from alsa, fixes #338919

2006-04-28 15:31:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
	  (gst_multi_fd_sink_remove_client_link):
	  * gst/tcp/gstmultifdsink.h:
	  Fix race condition in multifdsink that can lead to spurious
	  duplicate clients. this patch adds a new signal that is fired when
	  multifdsink has removed all references to the fd.
	  Fixes #339574.
	  Updated documentation.
	  API: client-fd-removed signal added

2006-04-28 15:24:00 +0000  Michael Smith <msmith@xiph.org>

	  gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
	  Original commit message from CVS:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
	  When asking g_value_array_new to prealloc elements, we may as well
	  ask for the right number of elements.

2006-04-28 15:08:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
	  (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
	  patch to make timestamp checking more tollerant to rounding
	  errors given that real discontinuities are to be marked on
	  buffers. Fixes some asf files and #338778.
	  Also avoid some crashers when we receive an event in the
	  NULL state.

2006-04-28 15:01:58 +0000  Michael Smith <msmith@xiph.org>

	  ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
	  (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
	  (gst_gnome_vfs_src_get_property),
	  (gst_gnome_vfs_src_send_additional_headers_callback),
	  (gst_gnome_vfs_src_received_headers_callback),
	  (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
	  (gst_gnome_vfs_src_stop):
	  * ext/gnomevfs/gstgnomevfssrc.h:
	  Remove ICY handling (mostly) from gnomevfssrc, in favour of
	  proper shared support within icydemux.

2006-04-28 14:49:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
	  Original commit message from CVS:
	  * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
	  (gst_video_rate_swap_prev), (gst_video_rate_chain):
	  fix up docs
	  fix a leak when no caps negotiated
	  fix counting of input frames
	  * tests/check/elements/.cvsignore:
	  * tests/check/elements/videorate.c: (assert_videorate_stats),
	  (GST_START_TEST), (videorate_suite):
	  add tests for these

2006-04-28 14:48:11 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
	  (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
	  (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
	  (gst_ring_buffer_pause), (gst_ring_buffer_stop),
	  (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
	  (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
	  (gst_ring_buffer_commit), (gst_ring_buffer_read),
	  (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
	  (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
	  Check arguments passed to public functions instead of
	  crashing.

2006-04-28 14:37:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
	  (gst_base_audio_src_get_time), (gst_base_audio_src_create):
	  GstBaseAudioSrc must be live or it does not work.
	  * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
	  Don't set live to TRUE as this is the default in the parentclass.

2006-04-28 14:37:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* win32/common/config.h:
	  update config.h
	  Original commit message from CVS:
	  update config.h

2006-04-28 14:33:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
	  Original commit message from CVS:
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
	  (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
	  Videoscale doesn't pass on pixel-aspect ratio. Handle all
	  fixation cases better. Fixes #338991

2006-04-28 14:24:38 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
	  Handle 0/1 framerate correctly Fixes #331901.

2006-04-28 14:22:16 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
	  Original commit message from CVS:
	  * tests/check/elements/audioconvert.c: (get_float_caps),
	  (GST_START_TEST), (audioconvert_suite):
	  Added check for correct clipping when doing float samples
	  in audioconvert.

2006-04-28 14:19:49 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videorate/gstvideorate.c: Print more debugging info.
	  Original commit message from CVS:
	  * gst/videorate/gstvideorate.c: (gst_video_rate_event),
	  (gst_video_rate_chain):
	  Print more debugging info.

2006-04-28 14:17:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
	  (resample_set_state_from_caps):
	  Add support for other formats audioresample can handle such as
	  32 bits in and float and 64 bits float. Fixes #301759

2006-04-28 14:12:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
	  Original commit message from CVS:
	  * gst/audioconvert/audioconvert.c: (float):
	  correctly clip float samples > 1.0. Fixes #338718

2006-04-28 13:35:34 +0000  Young-Ho Cha <ganadist@chollian.net>

	  ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
	  Original commit message from CVS:
	  Patch by: Young-Ho Cha  <ganadist at chollian net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
	  (gst_text_overlay_render_text):
	  Don't strip newlines from the text. Also, center lines
	  within multi-line paragraphs (#339405).

2006-04-28 12:15:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
	  Fix wavpack typefinding to work in more cases (don't peek
	  for chunks of multiple hundred kBs at once, but process
	  things step-by-step in smaller units). Fixes #339786.

2006-04-28 10:58:41 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.10.6 ===

2006-04-28 10:53:40 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.signals:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* docs/upload.mak:
	  releasing 0.10.6
	  Original commit message from CVS:
	  releasing 0.10.6

2006-04-28 10:42:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* win32/MANIFEST:
	* win32/common/config.h:
	  dist more win32 files
	  Original commit message from CVS:
	  dist more win32 files

2006-04-28 10:41:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-04-27 00:19:29 +0000  David Schleef <ds@schleef.org>

	  gst/videoscale/gstvideoscale.c: Add call to oil_init().
	  Original commit message from CVS:
	  * gst/videoscale/gstvideoscale.c: Add call to oil_init().
	  Fixes #338897.

2006-04-26 17:20:31 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* win32/common/config.h:
	  new prerelease
	  Original commit message from CVS:
	  new prerelease

2006-04-26 17:17:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
	  Original commit message from CVS:
	  2006-04-26  Thomas Vander Stichele  <thomas at apestaart dot org>
	  patch by: Wim Taymans
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_perform_seek):
	  make sure correct newsegments are sent, so that the decoder
	  and the demuxer agree on timestamps.  Fixes playback of a lot
	  of Ogg files that do not start from 0.  Fixes #339833.

2006-04-26 16:44:20 +0000  Edward Hervey <edward@fluendo.com>

	  Fix an infinite loop if frames are passed in with wrongly ordered timestamps.  Fixes #339013.
	  Original commit message from CVS:
	  Patch by: Edward Hervey  <edward@fluendo.com>
	  * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
	  * tests/check/Makefile.am:
	  * tests/check/elements/videorate.c: (assert_videorate_stats),
	  (setup_videorate), (cleanup_videorate), (GST_START_TEST),
	  (videorate_suite), (main):
	  Fix an infinite loop if frames are passed in with wrongly ordered
	  timestamps.  Fixes #339013.

2006-04-26 13:55:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	* win32/common/config.h:
	  prerelease
	  Original commit message from CVS:
	  prerelease

2006-04-22 21:25:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files.  Fixes #339212.
	  Original commit message from CVS:
	  Patch by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
	  fix typefinding on some ISO files.  Fixes #339212.

2006-04-22 21:19:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: add another H264 fourcc.  Fixes #339047.
	  Original commit message from CVS:
	  Patch by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  add another H264 fourcc.  Fixes #339047.

2006-04-22 21:12:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
	  Original commit message from CVS:
	  Patch by: Jan Schmidt
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_bufferalloc):
	  Restore old StreamSelector behaviour.
	  Fixes #338419.

2006-04-13 09:26:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* gst-libs/gst/rtp/Makefile.am:
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	  reverting rtp patches to fix freeze break on -base as explained on the list
	  Original commit message from CVS:
	  reverting rtp patches to fix freeze break on -base as explained on the list

2006-04-13 03:55:12 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
	  Original commit message from CVS:
	  2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
	  * gst-libs/gst/rtp/gstrtpbuffer.h:
	  Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
	  * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
	  New RTP audio base payloader class. Supports frame or sample based codecs

2006-04-12 11:04:53 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  update libtool versioning
	  Original commit message from CVS:
	  update libtool versioning

2006-04-12 10:58:00 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* win32/common/config.h:
	  prerelease
	  Original commit message from CVS:
	  prerelease

2006-04-11 17:31:29 +0000  Antoine Tremblay <hexa00@gmail.com>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
	  Original commit message from CVS:
	  Patch by: Antoine Tremblay  <hexa00 at gmail dot com>
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
	  Fix some memory leaks: on finalize, free buffers left in the queue
	  before destroying the queue; in _push(), unref rtp_buf even if
	  the process vfunc returned a NULL buffer as output buffer (#337548);
	  demote some recuring debug messages to LOG level.

2006-04-11 15:01:51 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  fix version number macro
	  Original commit message from CVS:
	  fix version number macro

2006-04-11 14:42:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: More cleanups.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
	  (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	  (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
	  (gst_ogg_demux_loop):
	  More cleanups.
	  Respect segment stop when emiting EOS or SEGMENT_DONE.
	  Fixes (#337945).

2006-04-11 10:45:32 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gststreamselector.c: Don't leak pad name.
	  Original commit message from CVS:
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_get_property):
	  Don't leak pad name.

2006-04-11 09:42:52 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	  Mention bug #336617 closed by recent commit
	  Original commit message from CVS:
	  Mention bug #336617 closed by recent commit

2006-04-10 20:32:46 +0000  Michael Smith <msmith@xiph.org>

	  tests/check/: so that FC4 buildslaves can pass.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  * tests/check/gst-plugins-base.supp:
	  Suppress an old libtheora bug (fixed in more recent versions), so
	  that FC4 buildslaves can pass.

2006-04-10 19:13:30 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Don't leak events.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
	  (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
	  (gst_ogg_demux_init), (gst_ogg_demux_finalize),
	  (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
	  (gst_ogg_demux_loop):
	  Don't leak events.
	  Remember what error we got when finding chains, if we
	  were shutdown, that would not be an error.

2006-04-10 17:05:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_event):
	  Starting the ringbuffer when we did not acquire it can cause
	  a deadlock, is pointless and causes nasty things for
	  subclasses.
	  Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.

2006-04-10 15:17:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Add some more debugging.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
	  (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
	  (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
	  (gst_ogg_demux_deactivate_current_chain),
	  (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
	  (gst_ogg_demux_bisect_forward_serialno),
	  (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
	  Add some more debugging.

2006-04-10 14:52:10 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* ext/theora/theoraenc.c:
	  fix width of docs
	  Original commit message from CVS:
	  fix width of docs

2006-04-10 10:29:21 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/theoradec.c: Some more debug info.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_dec_src_event),
	  (theora_handle_data_packet):
	  Some more debug info.
	  * tests/examples/seek/seek.c: (start_seek), (main):
	  Print element messages too.

2006-04-09 17:14:22 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
	  Original commit message from CVS:
	  * gst/audioresample/debug.h:
	  replace debug macros with variable number of parameters
	  by a simple alias to gstreamer standard debug macros
	  (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
	  supported by MSVC 6.0 and 7.1)
	  * gst/audioresample/resample.h:
	  define M_PI and rint for WIN32
	  * win32/common/libgstaudio.def:
	  * win32/common/libgstriff.def:
	  * win32/common/libgsttag.def:
	  * win32/common/libgstvideo.def:
	  add new exported functions
	  * win32/vs6:
	  update project files

2006-04-08 21:02:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixeroptions.c:
	  (gst_alsa_mixer_options_class_init):
	  * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
	  * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
	  * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
	  * gst-libs/gst/audio/gstaudiofilter.c:
	  (gst_audio_filter_class_init):
	  * gst-libs/gst/audio/gstaudiosink.c:
	  (gst_audioringbuffer_class_init):
	  * gst-libs/gst/audio/gstaudiosrc.c:
	  (gst_audioringbuffer_class_init):
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
	  * gst-libs/gst/interfaces/colorbalancechannel.c:
	  (gst_color_balance_channel_class_init):
	  * gst-libs/gst/interfaces/mixeroptions.c:
	  (gst_mixer_options_class_init):
	  * gst-libs/gst/interfaces/mixertrack.c:
	  (gst_mixer_track_class_init):
	  * gst-libs/gst/interfaces/tunerchannel.c:
	  (gst_tuner_channel_class_init):
	  * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
	  * gst-libs/gst/netbuffer/gstnetbuffer.c:
	  (gst_netbuffer_class_init):
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  (gst_basertppayload_class_init):
	  * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
	  * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
	  * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_class_init):
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
	  * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
	  * sys/v4l/gstv4lcolorbalance.c:
	  (gst_v4l_color_balance_channel_class_init):
	  * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
	  * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
	  * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
	  * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
	  (gst_v4l_tuner_norm_class_init):
	  * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
	  * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
	  Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)

2006-04-08 18:09:17 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Fix broken GObject macros
	  Original commit message from CVS:
	  * ext/pango/gsttextrender.h:
	  * gst-libs/gst/audio/gstaudiosink.h:
	  * gst-libs/gst/audio/gstaudiosrc.h:
	  * gst-libs/gst/audio/gstbaseaudiosink.h:
	  * gst-libs/gst/audio/gstbaseaudiosrc.h:
	  * gst-libs/gst/audio/gstringbuffer.h:
	  * gst-libs/gst/rtp/gstbasertpdepayload.h:
	  * gst-libs/gst/rtp/gstbasertppayload.h:
	  * gst-libs/gst/video/gstvideofilter.h:
	  * gst-libs/gst/video/gstvideosink.h:
	  * gst/playback/gstplaybasebin.h:
	  * gst/tcp/gstmultifdsink.h:
	  * sys/v4l/gstv4lelement.h:
	  Fix broken GObject macros

2006-04-08 16:21:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
	  More debug to trace why my USB headset is not working with gst

2006-04-07 17:18:11 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (group_destroy):
	  Clean up our group elements properly in the case where it never
	  got committed - it still got added unconditionally to the bin.

2006-04-07 15:14:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/theoradec.c: Unref unhandled events.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_dec_sink_event),
	  (theora_handle_data_packet), (theora_dec_chain):
	  Unref unhandled events.
	  Protect against empty buffers.
	  Perform QoS on running time.

2006-04-07 13:24:54 +0000  Michael Smith <msmith@xiph.org>

	  ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
	  (gst_vorbis_enc_chain):
	  Remove leaks from vorbisenc.
	  Mostly minor changes, the only significant one is that now the
	  buffers we set as 'streamheader' on the caps are copies of the
	  original buffers, to avoid circular refcounting problems.

2006-04-07 09:51:35 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
	  Don't remove our mute-probe if someone else already did so.
	  Don't set a 2nd one if there is already one pending on the pad.
	  * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
	  (do_playbin_seek):
	  When a seek fails, ensure that playbin is still set back to playing.
	  * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
	  (mpeg_ts_type_find), (plugin_init):
	  Add a typefind function for mpeg-ts streams.

2006-04-06 11:40:45 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* gst/audiotestsrc/gstaudiotestsrc.c:
	* gst/videorate/gstvideorate.c:
	  gst/videorate/gstvideorate.c (gst_video_rate_reset)
	  Original commit message from CVS:
	  2006-04-06  Andy Wingo  <wingo@pobox.com>
	  * gst/videorate/gstvideorate.c (gst_video_rate_reset)
	  (gst_video_rate_init): Caps-related parameters should not be reset
	  by a flush -- move their inits to the instance init function.
	  (gst_video_rate_flush_prev): Don't complain if gst_pad_push
	  is not OK, just return the result.
	  * gst/audiotestsrc/gstaudiotestsrc.c
	  (gst_audio_test_src_class_init)
	  (gst_audio_test_src_get_times): Re-enable is-live=true, as was
	  broken by Stefan's commit on 24 March.

2006-04-06 10:50:14 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
	  Original commit message from CVS:
	  2006-04-06  Andy Wingo  <wingo@pobox.com>
	  * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
	  buffers being pushed out. Fixes oggmux ! multifdsink.

2006-04-05 13:05:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
	  (gst_vorbis_dec_init), (vorbis_dec_finalize):
	  * ext/vorbis/vorbisdec.h:
	  * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
	  (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
	  (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
	  (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
	  (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
	  (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
	  (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
	  (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
	  (gst_vorbis_enc_buffer_from_packet),
	  (gst_vorbis_enc_buffer_from_header_packet),
	  (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
	  (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
	  (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
	  (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
	  (gst_vorbis_enc_change_state):
	  * ext/vorbis/vorbisenc.h:
	  Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
	  vorbisenc adhere to the official nomenclature; use boilerplate
	  macro.

2006-04-04 11:20:24 +0000  Andy Wingo <wingo@pobox.com>

	  gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
	  Original commit message from CVS:
	  2006-04-04  Andy Wingo  <wingo@pobox.com>
	  * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
	  Whoops, fix bug introduced. Bad hacker!

2006-04-04 11:15:00 +0000  Andy Wingo <wingo@pobox.com>

	  gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
	  Original commit message from CVS:
	  2006-04-04  Andy Wingo  <wingo@pobox.com>
	  * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
	  Properly handle the case where you get EOS before any buffers are
	  received. Use gst_buffer_make_metadata_writable where appropriate.

2006-04-04 10:16:46 +0000  Andy Wingo <wingo@pobox.com>

	  ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
	  Original commit message from CVS:
	  2006-04-04  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/theoradec.c (theora_handle_data_packet): This value
	  is often negative -- make it signed so as not to wrap around.
	  Fixes segfaults introduced on 9 March.

2006-04-03 16:43:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/: Don't try to store a gdouble in a gboolean.
	  Original commit message from CVS:
	  * ext/theora/gsttheoradec.h:
	  * ext/theora/theoradec.c: (theora_dec_src_event):
	  Don't try to store a gdouble in a gboolean.
	  Small cleanups.

2006-04-03 12:55:18 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggmux.c: Oggmux sucks.
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
	  Oggmux sucks.
	  Make it suck slightly less by writing out the final page.
	  Still can't encode a vorbis-in-ogg file correctly, though.

2006-04-03 08:49:06 +0000  Andy Wingo <wingo@pobox.com>

	  ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
	  Original commit message from CVS:
	  2006-04-03  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
	  a g_print.

2006-04-03 08:32:21 +0000  Andy Wingo <wingo@pobox.com>

	  ext/theora/theora.c (plugin_init): Register theoraparse.
	  Original commit message from CVS:
	  2006-04-03  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/theora.c (plugin_init): Register theoraparse.
	  * ext/theora/gsttheoraparse.h:
	  * ext/theora/theoraparse.c: New files implementing a theora
	  parser. Now we can properly remux ogg/theora+vorbis, yay.

2006-04-03 08:28:58 +0000  Andy Wingo <wingo@pobox.com>

	  ext/vorbis/vorbisparse.c: Add some docs and a copyright.
	  Original commit message from CVS:
	  2006-04-03  Andy Wingo  <wingo@pobox.com>
	  * ext/vorbis/vorbisparse.c: Add some docs and a copyright.

2006-04-01 15:34:38 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* configure.ac:
	  don't use AS_LIBTOOL_TAGS, it doesn't work
	  Original commit message from CVS:
	  don't use AS_LIBTOOL_TAGS, it doesn't work

2006-04-01 11:41:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* ext/pango/gsttextoverlay.c:
	* sys/v4l/gstv4lsrc.c:
	  remove BT8x8 from description, works for more devices
	  Original commit message from CVS:
	  remove BT8x8 from description, works for more devices

2006-04-01 11:21:30 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  Fixed the sample pipeline (see #323798)

2006-04-01 09:50:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  use AS_VERSION and AS_NANO more cleanups
	  Original commit message from CVS:
	  * configure.ac:
	  * win32/common/config.h:
	  * win32/common/config.h.in:
	  use AS_VERSION and AS_NANO
	  more cleanups

2006-03-31 17:08:41 +0000  Andy Wingo <wingo@pobox.com>

	  ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
	  Original commit message from CVS:
	  2006-03-31  Andy Wingo  <wingo@pobox.com>
	  * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
	  uninitialized variable return that would happen.

2006-03-31 16:57:47 +0000  Andy Wingo <wingo@pobox.com>

	  ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
	  Original commit message from CVS:
	  2006-03-31  Andy Wingo  <wingo@pobox.com>
	  * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
	  uninitialized variable return that would never happen.

2006-03-31 16:43:43 +0000  Andy Wingo <wingo@pobox.com>

	  ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
	  Original commit message from CVS:
	  2006-03-31  Andy Wingo  <wingo@pobox.com>
	  * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
	  (vorbis_parse_sink_event): Add an event function to flush our
	  state on a seek, and to drain buffers on a premature EOS.
	  (vorbis_parse_push_headers, vorbis_parse_clear_queue)
	  (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
	  (vorbis_parse_chain, vorbis_parse_queue_buffer)
	  (vorbis_parse_drain_queue): Queue up buffers until we can set
	  their timestamps and granulepos values.
	  * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
	  and keep track of data needed for deriving granulepos and
	  timestamps for buffers.

2006-03-30 11:05:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	  expose pluginsdir so gonlin can use it for tests
	  Original commit message from CVS:
	  expose pluginsdir so gonlin can use it for tests

2006-03-30 10:03:56 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base.pc.in:
	  add ccda to libraries
	  Original commit message from CVS:
	  add ccda to libraries

2006-03-29 14:00:08 +0000  j^ <j@bootlab.org>

	  better/unified long descriptions
	  Original commit message from CVS:
	  Patch by: j^ <j at bootlab dot org>
	  * ext/alsa/gstalsamixerelement.c:
	  (gst_alsa_mixer_element_class_init):
	  * ext/alsa/gstalsasink.c:
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
	  * ext/ogg/gstoggdemux.c:
	  * ext/ogg/gstoggmux.c:
	  * ext/ogg/gstoggparse.c:
	  * ext/pango/gstclockoverlay.c:
	  * ext/pango/gsttextoverlay.c:
	  * ext/pango/gsttextrender.c:
	  * ext/pango/gsttimeoverlay.c:
	  * ext/theora/theoradec.c:
	  * ext/theora/theoraenc.c:
	  * ext/vorbis/vorbisdec.c:
	  * ext/vorbis/vorbisenc.c:
	  * gst/audioconvert/gstaudioconvert.c:
	  * gst/subparse/gstsubparse.c:
	  * gst/tcp/gstmultifdsink.c:
	  * gst/tcp/gsttcpclientsink.c:
	  * gst/tcp/gsttcpclientsrc.c:
	  * gst/tcp/gsttcpserversink.c:
	  * gst/tcp/gsttcpserversrc.c:
	  better/unified long descriptions
	  Fixes #336477

2006-03-29 13:54:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
	  (stop_seek):
	  Don't let double and tripple clicks mess up our state.

2006-03-28 13:13:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gen_video_element),
	  (gen_text_element), (gen_audio_element), (gen_vis_element):
	  Error out gracefully when we can't create any of the usual
	  conversion elements for some reason. Also, don't try to
	  create an audioscale (sic) element that's not used anyway.

2006-03-28 10:21:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_source):
	  Don't post RESOURCE_NOT_FOUND error when we can't find a source
	  element for a particular protocol, that's confusing for users.
	  Instead, post a RESOURCE_FAILED error, so that our own error
	  message is actually shown in totem etc. (#336303).

2006-03-27 16:36:46 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
	  (gst_gnome_vfs_src_get_icy_metadata):
	  Fix some minor memory leaks (#336194).

2006-03-27 16:15:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfs.c:
	  (gst_gnome_vfs_location_to_uri_string):
	  * ext/gnomevfs/gstgnomevfs.h:
	  * ext/gnomevfs/gstgnomevfssink.c:
	  (gst_gnome_vfs_sink_set_property):
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
	  Make gnomevfssink accept filenames as well as URIs for the
	  "location" property, just like gnomevfssrc does (and
	  filesrc/filesink do) (#336190).

2006-03-24 20:35:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
	  Original commit message from CVS:
	  * tests/check/generic/clock-selection.c: (GST_START_TEST):
	  set to NULL before unreffing, fixes a valgrind leak.
	  Why was this not triggering the error that an object needs to
	  be NULL before unreffing ?
	  * win32/common/config.h:
	  update

2006-03-24 17:57:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (convert_encoding),
	  (gst_sub_parse_change_state):
	  * gst/subparse/gstsubparse.h:
	  Text subtitle files may or may not be UTF-8. If it's not, we
	  don't really want to see '?' characters in place of non-ASCII
	  characters like accented characters. So let's assume the input
	  is UTF-8 until we come across text that is clearly not. If it's
	  not UTF-8, we don't really know what it is, so try the following:
	  (a) see whether the GST_SUBTITLE_ENCODING environment variable
	  is set; if not, check (b) if the current locale encoding is
	  non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
	  the current locale encoding is UTF-8 and the environment variable
	  was not set to any particular encoding. Not perfect, but better
	  than nothing (and better than before, I think) (fixes #172848).

2006-03-24 17:39:45 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/plugins/tmpl/.gitignore:
	* tests/check/libs/.gitignore:
	* tests/check/pipelines/.gitignore:
	* tests/examples/volume/.gitignore:
	  ignore more
	  Original commit message from CVS:
	  ignore more

2006-03-24 17:26:54 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
	  Original commit message from CVS:
	  2006-03-24  Thomas Vander Stichele <thomas at apestaart dot org>
	  * configure.ac:
	  update core requirement to 0.10.4.1 because of async_playback
	  vmethod on GstBaseSink

2006-03-24 17:11:53 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  use DEBUG_FUNCPTR for collectpads
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
	  * gst/adder/gstadder.c: (gst_adder_init):
	  use DEBUG_FUNCPTR for collectpads

2006-03-24 14:11:20 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* Makefile.am:
	  don't go through check-torture if no check installed
	  Original commit message from CVS:
	  don't go through check-torture if no check installed

2006-03-24 10:42:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * ext/cdparanoia/gstcdparanoiasrc.c:
	  * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
	  (gst_gnome_vfs_sink_class_init):
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
	  * ext/ogg/gstoggmux.c:
	  * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
	  * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
	  (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
	  * ext/pango/gsttextoverlay.c:
	  * ext/pango/gsttextrender.c:
	  * ext/theora/theoradec.c:
	  * ext/theora/theoraenc.c:
	  * ext/vorbis/vorbisdec.c:
	  * ext/vorbis/vorbisenc.c:
	  * gst-libs/gst/audio/gstaudiofilter.c:
	  (gst_audio_filter_base_init):
	  * gst-libs/gst/audio/gstaudiofiltertemplate.c:
	  (gst_audio_filter_template_base_init):
	  * gst/adder/gstadder.c: (gst_adder_get_type):
	  * gst/adder/gstadder.h:
	  * gst/audioconvert/gstaudioconvert.c:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
	  (gst_audio_test_src_create):
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  * gst/playback/gstdecodebin.c:
	  * gst/playback/gstplaybin.c:
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_base_init):
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
	  * gst/volume/gstvolume.c:
	  * sys/v4l/gstv4lmjpegsink.c:
	  * sys/v4l/gstv4lmjpegsrc.c:
	  * tests/check/libs/cddabasesrc.c:
	  * tests/old/examples/gob/gst-identity2.gob:
	  Add docs for adder, use GST_ELEMENT_DETAILS macro,
	  define GstElementDetails at the top

2006-03-23 21:48:18 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
	  Original commit message from CVS:
	  * win32/common/libgstinterfaces.def:
	  Add a lot of export functions for gst-python
	  * win32/common/libgstinterfaces.dsp:
	  Add a missing include folder in the project configuration

2006-03-23 16:58:03 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_get_time), (gst_base_audio_src_create),
	  (gst_base_audio_src_change_state):
	  Fix audio sources, forgot to make the ringbuffer
	  startable...

2006-03-23 16:29:58 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_get_time), (gst_base_audio_src_create),
	  (gst_base_audio_src_change_state):
	  unparent instead of unref the ringbuffer.

2006-03-23 16:24:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
	  (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
	  Implement new async_play vmethod to start slaving and allow
	  playback start in case of async PLAY state changes.
	  * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
	  Enable QoS with new method in base class.

2006-03-23 11:18:19 +0000  Julien MOUTTE <julien@moutte.net>

	  gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
	  Original commit message from CVS:
	  Patch by: Julien MOUTTE <julien at moutte dot net>
	  * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
	  (gst_video_test_src_do_seek), (gst_video_test_src_create):
	  Partially handle 0 framerate, only EOS after the first frame
	  is missing.

2006-03-23 09:38:59 +0000  Fabrizio Gennari <fabrizio.ge@tiscali.it>

	  gst/: Patch for support of YVU9 AVI files (#334822)
	  Original commit message from CVS:
	  Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	  (gst_riff_create_video_template_caps):
	  * gst/ffmpegcolorspace/avcodec.h:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
	  (gst_ffmpegcsp_avpicture_fill):
	  * gst/ffmpegcolorspace/imgconvert.c:
	  Patch for support of YVU9 AVI files (#334822)

2006-03-22 15:29:25 +0000  Edward Hervey <bilboed@bilboed.com>

	  docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
	  Original commit message from CVS:
	  * docs/design/design-decodebin.txt:
	  Added design document for new decodebin
	  (Target Caps): text/x-pango-markup is also a default target caps.

2006-03-22 15:11:47 +0000  Edward Hervey <bilboed@bilboed.com>

	  docs/design/design-decodebin.txt: Added design document for new decodebin
	  Original commit message from CVS:
	  * docs/design/design-decodebin.txt:
	  Added design document for new decodebin

2006-03-22 12:33:09 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_dispose):
	  Since we _parent the ringbuffer, we also need to
	  _unparent instead of a plain _unref.

2006-03-22 12:28:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/seek.c: Add scrub checkbox.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
	  (stop_seek), (scrub_toggle_cb), (main):
	  Add scrub checkbox.

2006-03-21 17:47:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
	  Original commit message from CVS:
	  * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
	  (gst_ogg_parse_chain):
	  Fix very inefficient usage of linked lists (#335365).

2006-03-21 14:26:01 +0000  Edward Hervey <bilboed@bilboed.com>

	  gcc 4.1 unreferenced pointer fixes.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
	  * gst/playback/gstplaybin.c: (handoff):
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_set_property):
	  gcc 4.1 unreferenced pointer fixes.
	  * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
	  gst_buffer_ref() now takes a GstBuffer*.

2006-03-20 18:09:41 +0000  Julien Moutte <julien@moutte.net>

	  sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
	  Original commit message from CVS:
	  2006-03-20  Julien MOUTTE  <julien@moutte.net>
	  * sys/xvimage/xvimagesink.c:
	  (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
	  by Jan Schmidt.

2006-03-19 11:37:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
	  (id3v1_type_find), (apetag_type_find), (plugin_init):
	  Can't do tag preferences via probability, as tags would then
	  lose against types that are recognised with MAXIMUM probability
	  (like .wav); so let all tag typefinders return MAXIMUM themselves
	  and order them via the rank. Split ID3v1 and ID3v2 typefinders so
	  that we can prefer APE to ID3v1 (fixes #335028).

2006-03-17 17:48:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_change_state):
	  * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
	  (gst_ring_buffer_may_start):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Only start playback if we are playing.
	  should fix #330748.

2006-03-17 13:11:45 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Revert accidental commits to these files.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
	  * win32/common/config.h:
	  Revert accidental commits to these files.

2006-03-16 20:01:03 +0000  Michal Benes <michal.benes@xeris.cz>

	  tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
	  Original commit message from CVS:
	  Patch by: Michal Benes  <michal dot benes at xeris dot cz>
	  * tests/Makefile.am:
	  Don't try to build tests in tests/icles if we
	  don't have X (#323852)

2006-03-16 13:08:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c:
	  Add TXXX frame identifiers for replaygain stuff as used
	  by some taggers (see #323721).

2006-03-16 10:22:27 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
	  Original commit message from CVS:
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_set_property),
	  (gst_stream_selector_bufferalloc):
	  Preserve the existing buggy streamselector behaviour by performing
	  a fallback buffer allocation when downstream isn't linked yet.
	  This should really be fixed in playbin by blocking pads until it's
	  linked them.
	  Also, use gst_pad_alloc_buffer instead of
	  gst_pad_alloc_buffer_and_set.

2006-03-15 22:40:08 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c:
	  Don't crash on unknown ID3v2 TXXX frames.

2006-03-15 17:59:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
	  Chain up to the parent finalize method.
	  Add 32-bit sample size to the template caps.
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	  (gst_riff_create_video_template_caps):
	  Add the fourcc that the VMWare codec uses.
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_set_property),
	  (gst_stream_selector_bufferalloc),
	  (gst_stream_selector_request_new_pad):
	  For the active pad, forward buffer-alloc requests, otherwise
	  return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
	  having to memcpy every frame when used by playbin.
	  * gst/tcp/gstmultifdsink.c:
	  (gst_multi_fd_sink_handle_client_write):
	  Get negotiated caps from the sink pad, rather than the sink
	  pad's peer.

2006-03-15 17:11:34 +0000  Tommi Myöhänen <ext-tommi.myohanen@nokia.com>

	  ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
	  Original commit message from CVS:
	  Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
	  Don't forget to set src->callbacks_pushed to FALSE again when
	  popping them, otherwise re-activation in a different mode won't
	  work (#334620).

2006-03-15 11:30:29 +0000  Sebastien Moutte <sebastien@moutte.net>

	  gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
	  Original commit message from CVS:
	  Patch by: Sebastien Moutte  <sebastien moutte net>
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
	  (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
	  (gst_ffmpeg_smpfmt_to_caps):
	  Replace __VA_ARGS__ caps creation macros with varargs functions.
	  Makes things compile on MSVC (#320765), looks nicer, and we can
	  tell the compiler to check for the NULL terminator.

2006-03-14 15:13:04 +0000  Fabrizio Gennari <fabrizio.ge@tiscali.it>

	  gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
	  Original commit message from CVS:
	  Patch by: Fabrizio Gennari  <fabrizio dot ge at tiscali dot it>
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Make sure the buffer we copy into is really always big
	  enough, this time for real (#333488).

2006-03-14 13:16:49 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Add support for 24bpp DIB (#305279).

2006-03-14 11:11:59 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/: Re-enable QoS after the release.
	  Original commit message from CVS:
	  * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
	  * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
	  (gst_video_scale_init), (gst_video_scale_src_event):
	  Re-enable QoS after the release.
	  Rework videoscale to use the base class src_event handler.

2006-03-14 09:51:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: back to CVS.
	  Original commit message from CVS:
	  * configure.ac:
	  back to CVS.

=== release 0.10.5 ===

2006-03-13 19:50:04 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* win32/common/config.h:
	  releasing 0.10.5
	  Original commit message from CVS:
	  releasing 0.10.5

2006-03-13 17:28:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-03-13 11:17:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  Part of previous cdparanoiasrc docs fixes, forgot to commit.

2006-03-12 14:56:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Add cdparanoiasrc to docs.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * docs/plugins/gst-plugins-base-plugins.hierarchy:
	  Add cdparanoiasrc to docs.
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  More GstCddaBaseSrc docs.

2006-03-12 13:47:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add new API to libgsttag: gst_tag_from_id3_user_tag().
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
	  * gst-libs/gst/tag/tag.h:
	  Add new API to libgsttag: gst_tag_from_id3_user_tag().

2006-03-11 19:47:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  NULL-terminate array of mpeg4 video file extensions.
	  Fixes crash on PPC (#334226).

2006-03-11 16:40:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  (gst_gnome_vfs_src_check_get_range):
	  gnome_vfs_uri_is_local() alone is not a good indicator
	  whether we can operate in pull-mode with a specific URI,
	  as it returns FALSE for file:// URIs that point to an
	  NFS-mounted path. Be more conservative here: whitelist
	  local files, blacklist http URIs and use the old
	  mechanism for anything else (fixes #334216).

2006-03-10 19:15:34 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  configure.ac: back to trunk
	  Original commit message from CVS:
	  * configure.ac:
	  back to trunk

=== release 0.10.4 ===

2006-03-10 19:05:13 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* docs/upload.mak:
	* win32/common/config.h:
	  releasing 0.10.4
	  Original commit message from CVS:
	  releasing 0.10.4

2006-03-10 12:37:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
	  Original commit message from CVS:
	  * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
	  Disable max-lateness by setting it to -1 for now, so that
	  we can bed QoS stuff in thoroughly between now and the next
	  release.

2006-03-10 11:09:23 +0000  Fabrizio <fabrizio.ge@tiscali.it>

	  gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Make sure we don't read beyond the palette buffer in case of
	  broken or manipulated files (#333488, patch by: Fabrizio
	  Gennari)

2006-03-10 10:44:02 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
	  Fix for variable not initialized.

2006-03-09 19:02:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	* win32/common/config.h:
	  prereleasing
	  Original commit message from CVS:
	  prereleasing

2006-03-09 17:58:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libvisual/visual.c: Small cleanups.
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_get_type),
	  (gst_visual_src_setcaps), (gst_vis_src_negotiate),
	  (gst_visual_chain):
	  Small cleanups.
	  * ext/theora/gsttheoradec.h:
	  * ext/theora/theoradec.c: (gst_theora_dec_init),
	  (gst_theora_dec_reset), (_theora_granule_time),
	  (theora_dec_src_convert), (theora_dec_sink_convert),
	  (theora_dec_src_query), (theora_dec_src_event),
	  (theora_dec_sink_event), (theora_handle_comment_packet),
	  (theora_handle_header_packet), (theora_dec_push),
	  (theora_handle_data_packet), (theora_dec_chain),
	  (theora_dec_change_state):
	  Add simple QoS.

2006-03-09 17:50:59 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
	  (audiocast_register_listener), (gst_gnome_vfs_src_start):
	  Some cleanups.

2006-03-09 17:45:39 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
	  Don't try to activate NULL chains.

2006-03-09 16:30:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
	  Fix invalid memory access to region before peek'd data (#332964).

2006-03-09 15:05:03 +0000  Christophe Fergeau <teuf@gnome.org>

	  closes #333510.
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
	  * ext/pango/gsttextrender.c: (gst_text_render_init):
	  * gst/adder/gstadder.c: (gst_adder_init):
	  Don't leak padtemplates, patch by Christophe Fergeau,
	  closes #333510.

2006-03-09 12:56:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
	  Fix invalid memory access: make sure string passed to
	  regexec() is NUL-termianted.

2006-03-09 12:37:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
	  (mp3_type_find):
	  Refactor mpeg/audio typefinding to make it more maintainable
	  and easier to fine-tune. Make probing into middle of the file
	  work properly (fixes #333900, also see #152688).

2006-03-09 11:10:03 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c:
	  (utf8_type_find_have_valid_utf8_at_offset):
	  Remove part from previous commit that was bogus:
	  g_utf8_validate() does in fact not accept embedded
	  zeroes, so we don't need to check for those (thanks
	  to Mike for the hint).

2006-03-08 17:11:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c:
	  (utf8_type_find_count_embedded_zeroes),
	  (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
	  Make plain/text typefinder more conservative: firstly, check
	  for embedded zeroes, which are perfectly valid UTF-8 characters,
	  but also a fairly good sign that something is not a plain text
	  file; secondly, probe into the middle of the file if possible.
	  If we can't probe into the middle, limit the probability value
	  to be returned to TYPE_FIND_POSSIBLE (see #333900).

2006-03-08 11:34:45 +0000  Michael Smith <msmith@xiph.org>

	  gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Make typefind function name for mpeg4 video unique.

2006-03-08 09:53:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/libvisual/visual.c: Cleanups, post nice errors.
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_init),
	  (gst_visual_clear_actors), (gst_visual_dispose),
	  (gst_visual_reset), (gst_visual_src_setcaps),
	  (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
	  (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
	  (gst_visual_chain), (gst_visual_change_state):
	  Cleanups, post nice errors.
	  Handle sink and src events.
	  Implement simple QoS.
	  * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
	  Use new basesink methods to configure max-lateness.
	  Small doc update.
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
	  Debug statement cleanups.
	  * gst/volume/gstvolume.c: (gst_volume_class_init):
	  Simple cleanup.

2006-03-08 09:50:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
	  (gst_text_overlay_init), (gst_text_overlay_set_property),
	  (gst_text_overlay_get_property):
	  Revert API/ABI break from March 1. Keep 'halign' and 'valign'
	  as string type properties, but mark them deprecated. Add
	  'halignment' and 'valignment' properties that use enums
	  instead of strings.

2006-03-08 09:37:12 +0000  Fabrizio <fabrizio.ge@tiscali.it>

	  gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Allow palettes with less than 256 colours in AVI files
	  (#333488, patch by: Fabrizio Gennari).

2006-03-07 21:56:09 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
	  Original commit message from CVS:
	  2006-03-07  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
	  (gst_text_overlay_video_event): Fix wrong EOS handling on text
	  pad. We were releasing the queued text buffer when we should keep
	  it until video pad gets EOS or discard the text buffer because it's
	  too old. That was eating the last subtitle buffer. Add some more
	  debug.

2006-03-07 17:28:36 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
	  (gst_text_overlay_video_chain):
	  Fix invalid memory access (we can't access a buffer after it's been
	  pushed downstream without taking a reference); fix memory leak (if
	  there's no text to render, bail out before allocating stuff).

2006-03-07 15:08:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
	  (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
	  * ext/pango/gsttextoverlay.h:
	  If input is plain text, escape it before passing it to
	  pango_layout_set_markup().

2006-03-07 13:01:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
	  Don't ignore flow return from gst_pad_push().

2006-03-07 12:49:03 +0000  Christophe Fergeau <teuf@gnome.org>

	  Don't leak references returned by gst_pad_get_parent()
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_getcaps),
	  (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
	  (gst_vorbisenc_convert_sink):
	  * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
	  (gst_audio_duration_from_pad_buffer):
	  * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
	  (gst_audio_filter_chain):
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_setcaps):
	  * gst-libs/gst/video/video.c: (gst_video_frame_rate),
	  (gst_video_get_size):
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
	  Don't leak references returned by gst_pad_get_parent()
	  (#333663, based on patch by: Christophe Fergeau).

2006-03-06 20:52:25 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/gnomevfs/gstgnomevfssink.c: change location param details
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
	  change location param details
	  * gst/volume/gstvolume.c: (plugin_init):
	  correct plugin description

2006-03-06 20:07:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
	  (gst_gnome_vfs_src_check_get_range):
	  Override GstBaseSrc::check_get_range() in order to avoid opening
	  the resource just to check whether we can operate in pull-mode or
	  not - we can predict that pretty well from the URI alone. Should
	  fix problems with last.fm (#331690). (Requires latest core CVS).

2006-03-06 16:18:51 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
	  Original commit message from CVS:
	  * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
	  (gst_video_sink_class_init):
	  Throw away frames that are later than 20 ms.

2006-03-06 14:14:47 +0000  Fabrizio <fabrizio.ge@tiscali.it>

	  gst-libs/gst/riff/riff-media.c:
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).

2006-03-05 23:39:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/ogg/gstoggmux.c: put Theora BOS pages before others.  This hardcodes the Ogg/Theora I profile, but hey.
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
	  (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
	  put Theora BOS pages before others.  This hardcodes
	  the Ogg/Theora I profile, but hey.

2006-03-05 23:06:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/gstoggmux.c:
	  changed more than 5 lines
	  Original commit message from CVS:
	  changed more than 5 lines

2006-03-05 22:57:58 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
	  Original commit message from CVS:
	  ogg muxing of vorbis and theora now has pages ordered correctly again,
	  even with delays.
	  * ext/ogg/README:
	  updated with some examples
	  * ext/theora/theoraenc.c: (granulepos_to_timestamp),
	  (granulepos_add), (theora_buffer_from_packet):
	  * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
	  (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
	  (gst_vorbisenc_chain):
	  implement strategy from ext/ogg/README
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
	  (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
	  (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
	  (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
	  Fix muxer so that oggz-validate is happy with all streams;
	  except for no eos mark, and the BOS page ordering
	  * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
	  (check_buffer_granulepos):
	  * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
	  update tests to check for OFFSET being set as requested
	  fixed type of granulepos, it's not a ClockTime

2006-03-05 21:34:23 +0000  Julien Moutte <julien@moutte.net>

	  sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size         before returning it. (#3...
	  Original commit message from CVS:
	  2006-03-05  Julien MOUTTE  <julien@moutte.net>
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
	  (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
	  Check that the xvimage we are creating has a correct size         before returning it. (#314897)

2006-03-05 13:44:05 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Give id3 and ape tag typefinders a rank slightly higher
	  than PRIMARY to ensure they're always run before any of
	  the other typefinders (in particular wav and mp3) (#324186).

2006-03-05 13:08:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Add support for '3IVD' fourcc (#333403).

2006-03-04 14:35:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  configure.ac: Bump requirements to GStreamer CVS for the new error enum.
	  Original commit message from CVS:
	  * configure.ac:
	  Bump requirements to GStreamer CVS for the new error enum.
	  * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
	  Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
	  space left on the device (fixes #333352).

2006-03-03 23:53:50 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/vs6: add a project file for libgstvolume update the workspace
	  Original commit message from CVS:
	  * win32/vs6:
	  add a project file for libgstvolume
	  update the workspace

2006-03-03 15:26:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/README:
	* ext/ogg/gstoggmux.c:
	  debug updates
	  Original commit message from CVS:
	  debug updates

2006-03-03 15:22:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
	  Original commit message from CVS:
	  2006-03-03  Thomas Vander Stichele  <thomas at apestaart dot org>
	  * ext/theora/theoraenc.c: (theora_set_header_on_caps):
	  * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
	  (GST_START_TEST):
	  Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
	  Set IN_CAPS on header buffers

2006-03-02 18:23:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Add audioresample to docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  Add audioresample to docs.
	  * gst/audioconvert/gstaudioconvert.c:
	  Add revision date.
	  * gst/audioresample/gstaudioresample.c:
	  (gst_audioresample_base_init), (gst_audioresample_class_init),
	  (gst_audioresample_init), (gst_audioresample_dispose),
	  (audioresample_get_unit_size), (audioresample_transform_caps),
	  (resample_set_state_from_caps), (audioresample_transform_size),
	  (audioresample_set_caps), (audioresample_event),
	  (audioresample_do_output), (audioresample_transform),
	  (audioresample_pushthrough), (gst_audioresample_set_property),
	  (gst_audioresample_get_property), (plugin_init):
	  * gst/audioresample/gstaudioresample.h:
	  Added docs.
	  Small code cleanups.

2006-03-02 18:12:33 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/videorate/Makefile.am:
	  fix wim's commit
	  Original commit message from CVS:
	  fix wim's commit

2006-03-02 17:48:40 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/gstoggmux.c:
	  debug using the actual GstPad, that allows us to see the serialno in the padname
	  Original commit message from CVS:
	  debug using the actual GstPad, that allows us to see the serialno in the padname

2006-03-02 17:46:36 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Added videoscale to docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  Added videoscale to docs.
	  * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
	  (gst_video_rate_swap_prev), (gst_video_rate_event),
	  (gst_video_rate_chain):
	  Fix typo in docs.
	  * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
	  (gst_video_scale_init), (gst_video_scale_prepare_size),
	  (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
	  (gst_video_scale_fixate_caps), (gst_video_scale_transform):
	  * gst/videoscale/gstvideoscale.h:
	  Added docs, examples.
	  Some code cleanups.
	  Post errors instead of g_warning.

2006-03-02 17:30:57 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/gstoggmux.c:
	  clean up debug messages
	  Original commit message from CVS:
	  clean up debug messages

2006-03-02 17:15:38 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/gstoggmux.c:
	  extra debugging from older version, makes it easier to compare
	  Original commit message from CVS:
	  extra debugging from older version, makes it easier to compare

2006-03-02 17:04:55 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ext/ogg/gstoggmux.c:
	  some space cleanup and debug fixes
	  Original commit message from CVS:
	  some space cleanup and debug fixes

2006-03-02 16:47:34 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/: Added some more docs to libs and plugins.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * docs/libs/gst-plugins-base-libs.types:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  Added some more docs to libs and plugins.
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Document ringbuffer some more.
	  * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
	  (gst_video_rate_setcaps), (gst_video_rate_reset),
	  (gst_video_rate_init), (gst_video_rate_flush_prev),
	  (gst_video_rate_swap_prev), (gst_video_rate_event),
	  (gst_video_rate_chain), (gst_video_rate_change_state):
	  * gst/videorate/gstvideorate.h:
	  Fix videorate to use segments.
	  Make it work with 0/1 framerates (closes #331903)
	  Handle EOS correctly.
	  Added docs.

2006-03-02 13:13:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
	  Original commit message from CVS:
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
	  (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
	  (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
	  In state change function, first chain up to parent class,
	  then handle downwards state change stuff. Remove some
	  commented out cruft from 0.8 code.

2006-03-02 12:35:59 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
	  Original commit message from CVS:
	  * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
	  (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
	  (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
	  (gst_ogm_parse_chain):
	  Don't remove/re-add source pad if the new caps are the same as
	  the old caps anyway (#333042). When removing source pad, don't
	  unref it afterwards - we didn't ref it when adding. Sprinkle some
	  GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
	  after using gst_pad_get_parent(). Return downstream flow return
	  value in chain function.

2006-03-02 11:28:23 +0000  Wim Taymans <wim.taymans@gmail.com>

	  docs/plugins/: Fix hierarchy, added some more elements to the docs.
	  Original commit message from CVS:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins.args:
	  * docs/plugins/gst-plugins-base-plugins.hierarchy:
	  * docs/plugins/gst-plugins-base-plugins.interfaces:
	  * docs/plugins/gst-plugins-base-plugins.signals:
	  Fix hierarchy, added some more elements to the docs.
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_get_type):
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
	  Fix docs for ffmpegcolorspace.

2006-03-01 19:24:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
	  (apetag_type_find), (ape_type_find), (plugin_init):
	  Some typefinding fine-tuning:
	  - rank ID3/APE tags in order of preference via probabilities, so that
	  ID3v2 > APEv2 > APEv1 > ID3v1.
	  - three or four bytes don't really justify MAXIMUM probability,
	  change those to 'very likely' (musepack and monkeysaudio).

2006-03-01 18:25:18 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Added alsa docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * ext/alsa/gstalsamixer.c:
	  * ext/alsa/gstalsamixer.h:
	  * ext/alsa/gstalsamixerelement.c:
	  (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
	  * ext/alsa/gstalsamixerelement.h:
	  * ext/alsa/gstalsasink.c:
	  * ext/alsa/gstalsasink.h:
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
	  (gst_alsasrc_init):
	  * ext/alsa/gstalsasrc.h:
	  Added alsa docs.
	  Small code cleanups.

2006-03-01 17:52:45 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/Makefile.am: Dist new header too,
	  Original commit message from CVS:
	  * ext/theora/Makefile.am:
	  Dist new header too,

2006-03-01 17:39:28 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Fix some more docs.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * ext/gnomevfs/gstgnomevfssink.h:
	  * ext/gnomevfs/gstgnomevfssrc.h:
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
	  * ext/vorbis/vorbisdec.h:
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
	  * ext/vorbis/vorbisenc.h:
	  * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
	  (vorbis_parse_chain), (vorbis_parse_change_state):
	  * ext/vorbis/vorbisparse.h:
	  * gst/audioconvert/gstaudioconvert.h:
	  * gst/tcp/gsttcpserversink.h:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  * gst/videotestsrc/gstvideotestsrc.h:
	  * gst/volume/gstvolume.c:
	  * gst/volume/gstvolume.h:
	  Fix some more docs.
	  Added docs for vorbisdec and vorbisparse.
	  Fix vorbisparse.

2006-03-01 16:24:37 +0000  Wim Taymans <wim.taymans@gmail.com>

	  Updated/added documentation.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * ext/pango/gstclockoverlay.h:
	  * ext/pango/gsttextoverlay.h:
	  * ext/pango/gsttextrender.h:
	  * ext/pango/gsttimeoverlay.h:
	  * ext/theora/gsttheoradec.h:
	  * ext/theora/gsttheoraenc.h:
	  * ext/theora/theoradec.c:
	  * ext/theora/theoraenc.c:
	  * gst/audioconvert/gstaudioconvert.h:
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
	  * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
	  * gst/tcp/gstmultifdsink.h:
	  Updated/added documentation.
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
	  (gst_text_overlay_halign_get_type),
	  (gst_text_overlay_wrap_mode_get_type),
	  (gst_text_overlay_base_init), (gst_text_overlay_class_init),
	  (gst_text_overlay_init), (gst_text_overlay_set_property),
	  (gst_text_overlay_get_property):
	  Fix up properties to be enums instead of string to make bindings,
	  introspection and automatic GUI creation possible.
	  Add getters for the properties.

2006-02-28 21:21:07 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  added defines of M_PI and M_PI_2
	  * gst/ffmpegcolorspace/avcodec.h:
	  removed #include "stdint.h" for win32 as _stdint.h is
	  autogenerated to win32/common
	  * win32/common/libgstaudio.def:
	  * win32/common/libgsttag.def:
	  added some exports
	  * win32/vs6:
	  some project files bugs corrected
	  * win32/vs7:
	  project files are reset to the default vs7 configuration
	  (they link to msvcr71.dll using default optimizations)

2006-02-28 19:08:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
	  Fix some docs.

2006-02-28 13:52:04 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
	  Original commit message from CVS:
	  * ext/alsa/gstalsasrc.c:
	  Set proper class on the ElementDetails:
	  Source/Audio instead of Src/Audio

2006-02-28 12:19:11 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
	  Original commit message from CVS:
	  * gst/videoscale/vs_scanline.c:
	  (vs_scanline_resample_nearest_RGBA):
	  Revert optimization in videoscale. It should go in liboil and have
	  an appropriate liboil function.

2006-02-28 11:06:24 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_provide_clock):
	  Don't try to provide a clock in the NULL state.

2006-02-28 11:04:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
	  (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
	  (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_deactivate_current_chain),
	  (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
	  (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
	  (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
	  (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
	  Use GstSegment infrastructure to remove duplicated code
	  and handle more seek cases correctly.

2006-02-28 10:39:19 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_transform):
	  Don't ignore return code from ffmpeg convert function.
	  * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
	  Split out some long statements to ease debugging.

2006-02-27 12:08:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (gst_visual_init),
	  (gst_vis_src_negotiate), (get_buffer), (plugin_init):
	  Don't use gst_pad_use_fixed_caps, because it prevents downstream from
	  being able to renegotiate the size. Instead, use the negotiation
	  algorithm from the goom plugin to pick an initial output caps.
	  Also, allow theoretical libvisual plugins that might support non-GL
	  output even if they also do GL.

2006-02-26 21:05:46 +0000  Julien Moutte <julien@moutte.net>

	  ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
	  Original commit message from CVS:
	  2006-02-26  Julien MOUTTE  <julien@moutte.net>
	  * ext/libvisual/visual.c: (gst_visual_init),
	  (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
	  (plugin_init): Load only non GL plugins. Fix some memleaks and
	  possible negotiation issues.

2006-02-24 23:19:44 +0000  Julien Moutte <julien@moutte.net>

	  gst-libs/gst/tag/tag.h: Adding Annodex tags here.
	  Original commit message from CVS:
	  2006-02-25  Julien MOUTTE  <julien@moutte.net>
	  * gst-libs/gst/tag/tag.h: Adding Annodex tags here.

2006-02-24 18:55:27 +0000  Michael Smith <msmith@xiph.org>

	  gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
	  (cmml_type_find), (plugin_init):
	  Fix CMML type find function to not require a specific minor version
	  of the CMML header.
	  Add an MPEG4 video elementary stream typefind function.

2006-02-24 17:31:53 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
	  (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
	  (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
	  (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
	  (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
	  (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
	  Annodex support in ogg demuxer. Doesn't do very much without the
	  other annodex patches (to come).

2006-02-24 16:21:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c:
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
	  Pick up palette for MS video v1 (#327028, patch by:
	  Fabrizio Gennari <fabrizio dot get at tiscali dot it>)

2006-02-24 13:54:04 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_caps_remove_format_info),
	  (gst_ffmpegcsp_get_unit_size):
	  The 'palette_data' field from incoming RGB caps shouldn't be
	  proxied on outgoing YUV caps; also, restrict unit size
	  adjustment in case of paletted data only to the unit that
	  actually has a palette. Fixes #330711.

2006-02-24 12:18:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
	  (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
	  (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
	  (gst_ffmpegcsp_get_unit_size):
	  Plug some memory leaks.

2006-02-24 10:18:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
	  Original commit message from CVS:
	  * sys/ximage/Makefile.am:
	  * sys/xvimage/Makefile.am:
	  Add some _CFLAGS and _LIBS that seem to be missing
	  and/or required for Cygwin (see #317048).

2006-02-24 00:07:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	* ChangeLog:
	  ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
	  Original commit message from CVS:
	  ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15

2006-02-22 18:46:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasrc.c:
	  Fix description as pointed out by caugier.

2006-02-22 10:29:22 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
	  Original commit message from CVS:
	  Reviewed by : Edward Hervey  <edward@fluendo.com>
	  * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
	  (qt_type_find):
	  Better 3gp typefinding.

2006-02-21 12:16:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
	  Don't send EOS event here, the base class will send one for us.
	  * gst/playback/gstplaybasebin.c: (prepare_output):
	  Subpictures without video stream aren't allowed either.
	  * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
	  Fix debug statement copy'n'paste-o.

2006-02-21 12:05:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
	  Original commit message from CVS:
	  * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
	  Fix issues with mixer keeping state when muting/unmuting
	  and when changing the volume whilst muted (see #331763
	  and #331765).

2006-02-20 18:27:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
	  (parse_subrip), (gst_sub_parse_format_autodetect):
	  Set right caps given that we send escaped text. Also,
	  honour <i></i>, <b></b> and <u></u> markers that can be found
	  in .srt files (fixes #310202).

2006-02-20 16:21:14 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/mixerutils.c:
	  (element_factory_rank_compare_func):
	  Make order in which elements are tried more determinable.

2006-02-20 15:57:51 +0000  Julien Moutte <julien@moutte.net>

	  gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
	  (remove_element_chain), (cleanup_decodebin),
	  (gst_decode_bin_change_state): Make decodebin reusable by
	  fixing remove_element_chain first and then introduce a
	  cleaner in state change to ->NULL. (Closes #331678)
	  ------------------------------------------------------

2006-02-19 14:32:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
	  use 0666 mask when creating files so umask gets applied
	  correctly. Fixes #331295.

2006-02-19 14:16:16 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
	  Original commit message from CVS:
	  * gst/subparse/Makefile.am:
	  * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
	  (gst_ssa_parse_dispose), (gst_ssa_parse_init),
	  (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
	  (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
	  (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
	  (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
	  * gst/subparse/gstssaparse.h:
	  * gst/subparse/gstsubparse.c: (plugin_init):
	  Add very basic parser for SSA subtitle streams (as often
	  found in matroska files).

2006-02-19 14:09:40 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (mimetype_is_raw):
	  That should be text/x-pango-markup, not text/x-pango-layout.

2006-02-19 12:41:03 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.c: Polishing.
	  Original commit message from CVS:
	  2006-02-19  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
	  Polishing.

2006-02-19 12:05:23 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.c: Fix state change deadlock.
	  Original commit message from CVS:
	  2006-02-19  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
	  (gst_text_overlay_finalize), (gst_text_overlay_init),
	  (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
	  (gst_text_overlay_render_text),
	  (gst_text_overlay_text_pad_link),
	  (gst_text_overlay_text_event), (gst_text_overlay_video_event),
	  (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
	  (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
	  Fix state change deadlock.

2006-02-19 11:56:28 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
	  Original commit message from CVS:
	  2006-02-19  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
	  (gst_text_overlay_finalize), (gst_text_overlay_init),
	  (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
	  (gst_text_overlay_render_text),
	  (gst_text_overlay_text_pad_link),
	  (gst_text_overlay_text_event), (gst_text_overlay_video_event),
	  (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
	  (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
	  * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
	  and subtitles files.

2006-02-19 00:40:38 +0000  Julien Moutte <julien@moutte.net>

	  gst/playback/gstdecodebin.c: pango layout should be considered as row.
	  Original commit message from CVS:
	  2006-02-19  Julien MOUTTE  <julien@moutte.net>
	  * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
	  should be considered as row.

2006-02-19 00:25:16 +0000  Julien Moutte <julien@moutte.net>

	  gst/playback/gststreaminfo.*: Introduce language informations.
	  Original commit message from CVS:
	  2006-02-19  Julien MOUTTE  <julien@moutte.net>
	  * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
	  (cb_probe):
	  * gst/playback/gststreaminfo.h: Introduce language informations.

2006-02-18 22:41:31 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
	  (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
	  Set shared memory segments to be deleted as soon as we have attached,
	  that way they get cleaned up automatically if we crash.

2006-02-18 19:53:48 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/: Those functions are called with lock held.
	  Original commit message from CVS:
	  2006-02-18  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
	  * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
	  functions are called with lock held.

2006-02-18 19:51:47 +0000  Julien Moutte <julien@moutte.net>

	* ChangeLog:
	  Forgot Changelog.
	  Original commit message from CVS:
	  Forgot Changelog.

2006-02-18 19:10:35 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
	  Original commit message from CVS:
	  2006-02-18  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
	  (gst_text_overlay_finalize), (gst_text_overlay_init),
	  (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
	  (gst_text_overlay_render_text),
	  (gst_text_overlay_text_pad_link),
	  (gst_text_overlay_text_pad_unlink),
	  (gst_text_overlay_text_event),
	  (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
	  (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
	  (gst_text_overlay_change_state): Refactoring of textoverlay
	  without collectpads. This now supports sparse subtitles coming
	  from a demuxer instead of a sub file. Seeking is still broken
	  though. Need to discuss with wtay some more on how to handle
	  seeking correctly.
	  * ext/pango/gsttextoverlay.h:
	  * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
	  subtitles coming from the demuxer.

2006-02-17 19:31:12 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisenc.c: Use some more scaling functions.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
	  (gst_vorbisenc_convert_sink):
	  Use some more scaling functions.

2006-02-17 16:12:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
	  Original commit message from CVS:
	  * ext/cdparanoia/gstcdparanoiasrc.c:
	  (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
	  (gst_cd_paranoia_paranoia_callback),
	  (gst_cd_paranoia_src_signal_is_being_watched),
	  (gst_cd_paranoia_src_read_sector):
	  * ext/cdparanoia/gstcdparanoiasrc.h:
	  Add back 'transport-error' and 'uncorrected-error' signals and
	  make them actually be fired when bad stuff happens (#319340).

2006-02-17 14:07:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
	  (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
	  (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
	  (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
	  (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
	  (gst_ring_buffer_pause), (gst_ring_buffer_stop),
	  (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
	  (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
	  (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
	  (gst_ring_buffer_clear):
	  Small cleanups.
	  Added some G_LIKELY.

2006-02-17 10:15:52 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/TODO: Update TODO
	  Original commit message from CVS:
	  * gst-libs/gst/audio/TODO:
	  Update TODO
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_get_offset):
	  When trying to play samples ASAP and we don't have a
	  previous sample, try to play at position 0 instead of
	  an invalid position.

2006-02-17 09:24:56 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_open),
	  (gst_alsasink_reset):
	  Also release lock when we get an error in _reset();
	  fix an error message.

2006-02-16 21:01:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
	  (gst_alsasink_init), (get_channel_free_structure),
	  (caps_add_channel_configuration), (gst_alsasink_getcaps),
	  (gst_alsasink_close):
	  * ext/alsa/gstalsasink.h:
	  Add support for more than 2 channels (#326720).

2006-02-16 20:19:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
	  Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
	  with 4 or 6 channels, assume a default channel layout to make things
	  work (not sure there's anything else we can do in those cases).

2006-02-16 19:18:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/multichannel.c: Minor docs fix.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/multichannel.c:
	  Minor docs fix.
	  * gst-libs/gst/riff/Makefile.am:
	  * gst-libs/gst/riff/riff-ids.h:
	  * gst-libs/gst/riff/riff-media.c:
	  (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
	  Add support for WAVEFORMATEX, eg. PCM audio with more than two
	  channels and a channel layout map.

2006-02-16 17:06:46 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
	  Original commit message from CVS:
	  Reviewed by Edward Hervey  <edward@fluendo.com>
	  * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
	  C-level optimization of the RGBA nearest neighbour function.
	  Eventually this might end up in liboil with vectorized versions.

2006-02-16 11:44:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/multichannel.c:
	  (gst_audio_get_channel_positions):
	  When we have more than 2 channels, but no channel layout is
	  specified in the caps, return some default channel layout
	  to the caller and warn about about a possibly buggy element
	  (could be buggy filtercaps as well of course) (#317038).

2006-02-16 09:29:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
	  Original commit message from CVS:
	  * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	  Add gst-libs/gst/cdda to list of lib search paths.

2006-02-15 12:20:47 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
	  Original commit message from CVS:
	  2006-02-15  Andy Wingo  <wingo@pobox.com>
	  * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
	  timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
	  to the Lord Jesus that I do not have to touch the ogg muxer ever
	  again.

2006-02-15 12:07:57 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
	  quicktime movie files can also contain 'uuid' atoms.

2006-02-14 18:52:52 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
	  Original commit message from CVS:
	  * gst/audioconvert/plugin.c: (plugin_init):
	  Register the GstAudioChannelPosition enum type with the type
	  system in the plugin_init function, so that it is known before
	  any element actually makes use of multi-channel stuff. This is
	  required for example if one wants to be able to deserialise/use
	  a caps string with channel positions before any pipeline has
	  been setup and started, like with gst-launch.

2006-02-14 13:45:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c:
	  (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
	  (gst_ring_buffer_samples_done), (wait_segment),
	  (gst_ring_buffer_commit), (gst_ring_buffer_clear):
	  Add some compiler G_(UN_)LIKELY help.
	  SIGNAL the ringbuffer waiters when going to PAUSED as well to
	  make sure they can exit their functions. Should fix #330748

2006-02-13 20:49:07 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  Windows does not have long long; copy the generated _stdint.h
	  Original commit message from CVS:
	  * Makefile.am:
	  * configure.ac:
	  * win32/MANIFEST:
	  * win32/common/_stdint.h:
	  Windows does not have long long; copy the generated _stdint.h
	  * win32/common/interfaces-enumtypes.c:
	  (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
	  (gst_mixer_track_flags_get_type),
	  (gst_tuner_channel_flags_get_type):
	  * win32/common/multichannel-enumtypes.c:
	  (gst_audio_channel_position_get_type):
	  update

2006-02-13 18:49:02 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
	  (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
	  Always sync on first sample we receive when starting.

2006-02-13 15:59:48 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/playback/gstplaybin.c: Update vis bin docs.
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gen_vis_element):
	  Update vis bin docs.
	  Move queue after tee so we don't queue video buffers but
	  audio samples instead. Fixes problems where the video queue
	  is filled and the audio queue empty.

2006-02-13 15:17:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
	  No need to push an EOS event here, GstBaseSrc will do that for us
	  when we return FLOW_UNEXPECTED.

2006-02-12 14:54:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
	  (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
	  (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
	  Use scale functions when possible.
	  Fix error messages.
	  Free clockid when after waiting for EOS.
	  Use G_(UN_)LIKLY when it makes sense.
	  Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.

2006-02-12 14:26:55 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (prepare_output):
	  Remove stray semi-colon (fixes #330888).

2006-02-11 23:35:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  sys/: Fix up the XShm call testing so that we catch errors, and don't cause new ones by attempting to detach from a s...
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
	  Fix up the XShm call testing so that we catch errors, and don't
	  cause new ones by attempting to detach from a segment we failed
	  to attach to. Fixes #312439.

2006-02-10 11:29:55 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (plugin_init):
	  Added flv file typefind (video/x-flv).

2006-02-10 10:53:33 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	  (gst_riff_create_video_template_caps):
	  Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
	  Also added the caps to the default set of riff video caps.

2006-02-09 19:05:23 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
	  Original commit message from CVS:
	  2006-02-09  Andy Wingo  <wingo@pobox.com>
	  * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
	  time and the end time of the last packet in the page.
	  (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
	  on the pages in our queue, set the duration as well. Reflow a
	  debug statement.
	  (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
	  Fixes bad muxing order.

2006-02-09 17:04:18 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
	  Original commit message from CVS:
	  * gst-libs/gst/rtp/gstbasertppayload.c:
	  (gst_basertppayload_setcaps), (gst_basertppayload_push):
	  update seqnum before setting it on the packet; this makes sure
	  that the timestamp and seqnum properties match after pushing
	  a buffer

2006-02-09 12:16:35 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	  changelog foo
	  Original commit message from CVS:
	  changelog foo

2006-02-09 11:46:03 +0000  Andy Wingo <wingo@pobox.com>

	* gst-libs/gst/audio/gstringbuffer.c:
	* win32/common/config.h:
	  kapowpowpow
	  Original commit message from CVS:
	  kapowpowpow

2006-02-09 11:36:18 +0000  Andy Wingo <wingo@pobox.com>

	  gst-libs/gst/audio/gstringbuffer.c
	  Original commit message from CVS:
	  2006-02-09  Andy Wingo  <wingo@pobox.com>
	  * gst-libs/gst/audio/gstringbuffer.c
	  (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
	  overflow after 13.5 hours of recording. Kapow!
	  * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
	  the buffer size -- we don't care about underrun/overrun reporting
	  right now, just need to return a useful value.

2006-02-09 11:21:33 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Back to CVS
	  Original commit message from CVS:
	  * configure.ac:
	  Back to CVS

=== release 0.10.3 ===

2006-02-09 11:18:22 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* win32/common/config.h:
	  Releasing 0.10.3
	  Original commit message from CVS:
	  Releasing 0.10.3

2006-02-08 18:37:38 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  configure.ac: Drat. Bump libtool version number for new API.
	  Original commit message from CVS:
	  * configure.ac:
	  Drat. Bump libtool version number for new API.
	  Prelease 0.10.2.3 (of 0.10.3)

2006-02-08 15:57:53 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  0.10.2.2 prerelease (of 0.10.3).
	  Original commit message from CVS:
	  * configure.ac:
	  * win32/common/config.h:
	  0.10.2.2 prerelease (of 0.10.3).

2006-02-08 15:50:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
	  Original commit message from CVS:
	  * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
	  Revert Andy's newsegment change pending a more correct
	  fix.

2006-02-08 12:46:14 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update .po files
	  Original commit message from CVS:
	  Update .po files

2006-02-08 11:04:09 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst/tcp/gstmultifdsink.c:
	  doc fixes
	  Original commit message from CVS:
	  doc fixes

2006-02-08 09:20:23 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
	  Original commit message from CVS:
	  :
	  * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
	  (qt_type_find), (plugin_init):
	  detect more files as 3gp
	  group and reorder the iso file formats

2006-02-07 18:32:00 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
	  Original commit message from CVS:
	  * ext/vorbis/vorbis.c: (plugin_init):
	  Register musicbrainz tags, so apps don't have to.

2006-02-07 17:44:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
	  (gst_tag_to_vorbis_tag):
	  Make sure we called gst_tag_register_musicbrainz_tags()
	  before possibly mapping a vorbiscomment string from/to a
	  musicbrainz tag.

2006-02-07 16:16:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
	  In case we can't find the required number of consecutive
	  mpeg audio frames to positively identify an MPEG audio
	  stream, check if there's at least a valid mpeg audio
	  frame right at offset 0 and if so suggest mpeg/audio
	  caps with a very low probability (#153004).

2006-02-07 15:52:26 +0000  Andy Wingo <wingo@pobox.com>

	  gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
	  Original commit message from CVS:
	  2006-02-07  Andy Wingo  <wingo@pobox.com>
	  * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
	  a TIME segment if we get timestamped buffers. Requires recent
	  fixes in core to work properly.

2006-02-07 14:57:46 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (prepare_output):
	  Don't print the URI as part of the error message, it
	  makes error dialogs look rather ugly, especially if
	  the URI is very long or has characters in it that
	  need escaping.

2006-02-07 13:11:31 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (prepare_output):
	  Error out if we have only text or subtitles, but nothing
	  else. Also error out if we have subtitles but no video
	  stream.

2006-02-07 11:44:39 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
	  Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
	  Post an error message on the bus when we encounter an
	  error, which will hopefully be more meaningful than the
	  'Internal Flow Error' message users get to see if we
	  just return GST_FLOW_ERROR.

2006-02-07 11:28:04 +0000  Andy Wingo <wingo@pobox.com>

	  configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
	  Original commit message from CVS:
	  2006-02-07  Andy Wingo  <wingo@pobox.com>
	  * configure.ac (GST_MAJORMINOR): Update core version req to
	  0.10.2.2, for the collectpads API addition (#330244).

2006-02-06 19:09:26 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
	  Return FALSE from plugin_init() when GnomeVFS can't
	  be initialised for some reason (#328423).

2006-02-06 13:26:54 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
	  Original commit message from CVS:
	  2006-02-06  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
	  Stick to seeking theory until i find the bug.
	  * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.

2006-02-06 12:38:48 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Make theoraenc and the tests leak free. Like, really.
	  Original commit message from CVS:
	  * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
	  (theora_enc_finalize), (theora_enc_sink_setcaps),
	  (theora_set_header_on_caps), (theora_enc_chain),
	  (theora_enc_change_state):
	  * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
	  Make theoraenc and the tests leak free. Like, really.

2006-02-05 23:31:05 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
	  Original commit message from CVS:
	  (theora_enc_finalize), (theora_enc_sink_setcaps):
	  Add a finalize method to ensure we clean up state even if
	  someone omitted the state change back to NULL.
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
	  (gst_vorbisenc_chain):
	  Free some more leaked bits.
	  * tests/check/pipelines/theoraenc.c: (start_pipeline),
	  (stop_pipeline):
	  Wait for state changes to happen if they're ASYNC.
	  This ought to teach those fancy pants buildbots a lesson.

2006-02-05 22:47:41 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstid3tag.c:
	  Add mapping for ID3 International Standard Recording Code
	  tag "TSRC"

2006-02-05 22:44:55 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/vorbis/vorbisenc.c: Don't leak tag names.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
	  Don't leak tag names.

2006-02-05 18:22:01 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Split libgsttag docs into multiple sections.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/tag/gstid3tag.c:
	  * gst-libs/gst/tag/gstvorbistag.c:
	  * gst-libs/gst/tag/tags.c:
	  Split libgsttag docs into multiple sections.

2006-02-05 18:01:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add libgsttag to the docs.
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
	  * gst-libs/gst/tag/gstvorbistag.c:
	  * gst-libs/gst/tag/tag.h:
	  * gst-libs/gst/tag/tags.c:
	  Add libgsttag to the docs.

2006-02-05 17:21:23 +0000  Julien Moutte <julien@moutte.net>

	  ext/pango/gsttextoverlay.c: Fix clockoverlay.
	  Original commit message from CVS:
	  2006-02-05  Julien MOUTTE  <julien@moutte.net>
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
	  (gst_text_overlay_init), (gst_text_overlay_src_event),
	  (gst_text_overlay_collected): Fix clockoverlay.

2006-02-05 17:15:17 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
	  Original commit message from CVS:
	  * docs/libs/compiling.sgml:
	  Fix typo: it's pkg-config, not pkg-gconfig
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * docs/libs/tmpl/gstgconf.sgml:
	  There is no libgstgconf in 0.10, remove it
	  from the docs.

2006-02-05 16:03:48 +0000  Julien Moutte <julien@moutte.net>

	  docs/libs/tmpl/gstcolorbalance.sgml: Updated.
	  Original commit message from CVS:
	  2006-02-05  Julien MOUTTE  <julien@moutte.net>
	  * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
	  (gst_text_overlay_src_event), (gst_text_overlay_collected):
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
	  (gst_sub_parse_class_init), (gst_sub_parse_init),
	  (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
	  (parse_mpsub), (parser_state_init), (handle_buffer),
	  (gst_sub_parse_chain), (gst_sub_parse_sink_event),
	  (plugin_init):
	  * gst/subparse/gstsubparse.h: Introduce seeking code.

2006-02-05 15:14:06 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
	  Original commit message from CVS:
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  Add comment about LANGUAGE tag inconsistency (we want
	  ISO-639-1, but extract three-letter identifiers?)
	  * po/POTFILES.in:
	  Add two translatable files.

2006-02-05 14:59:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
	  Original commit message from CVS:
	  * gst-libs/gst/tag/Makefile.am:
	  * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
	  * gst-libs/gst/tag/tag.h:
	  * gst-libs/gst/tag/tags.c:
	  (gst_tag_register_musicbrainz_tags_internal),
	  (gst_tag_register_musicbrainz_tags):
	  Forward-port some tags stuff from the 0.8 branch. This is
	  mostly the addition of musicbrainz tags and their mapping
	  to vorbistags, and a vorbistag mapping of the language tag.

2006-02-05 12:06:25 +0000  Julien Moutte <julien@moutte.net>

	  gst/playback/gstplaybin.c: Fix broken code refactoring.
	  Original commit message from CVS:
	  2006-02-05  Julien MOUTTE  <julien@moutte.net>
	  * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
	  refactoring.

2006-02-05 03:05:41 +0000  David Schleef <ds@schleef.org>

	  Add Dirac typefinding and add dirac format to oggmux.
	  Original commit message from CVS:
	  * ext/ogg/gstoggmux.c:
	  * gst/typefind/gsttypefindfunctions.c:
	  Add Dirac typefinding and add dirac format to oggmux.

2006-02-04 07:49:03 +0000  Michael Smith <msmith@xiph.org>

	* configure.ac:
	  Improve error message for liboil missingness.
	  Original commit message from CVS:
	  Improve error message for liboil missingness.

2006-02-03 19:23:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (try_to_link_1):
	  Don't put essential function call into
	  g_return_*() macro, otherwise it'll all be
	  replaced by NOOPs when compiling with
	  G_DISABLE_CHECKS defined.

2006-02-03 17:45:44 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggparse.c:
	* gst/tcp/gsttcpserversink.c:
	* sys/v4l/v4lsrc_calls.c:
	* sys/v4l/v4lsrc_calls.h:
	  Just make it compile with --disable-gst-debug.
	  Original commit message from CVS:
	  Just make it compile with --disable-gst-debug.

2006-02-03 12:51:47 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
	  (gst_alsasink_class_init), (gst_alsasink_init),
	  (gst_alsasink_write), (gst_alsasink_reset):
	  * ext/alsa/gstalsasink.h:
	  Add lock to protect alsa calls.
	  Implement reset to flush samples ASAP, does not work
	  with dmix though.

2006-02-02 18:18:31 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_provide_clock):
	  Ugh.. getting late I guess...

2006-02-02 18:13:26 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_provide_clock),
	  (gst_base_audio_sink_set_property),
	  (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
	  Don't try to provide a clock when we are not negotiated since
	  we might not be able to make it run.

2006-02-02 17:51:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (try_to_link_1):
	  Unlinking two source pads is ... hard.

2006-02-02 12:14:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/TODO: Updated.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/TODO:
	  Updated.
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
	  On EOS, wait till the last sample is played before posting EOS.

2006-02-02 08:53:27 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* tests/check/pipelines/theoraenc.c:
	  comment on my understanding
	  Original commit message from CVS:
	  comment on my understanding

2006-02-02 08:47:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* common:
	* tests/check/pipelines/theoraenc.c:
	  reformat to fit 80 chars
	  Original commit message from CVS:
	  reformat to fit 80 chars

2006-02-02 00:04:37 +0000  Kai Vehmanen <kv2004@eca.cx>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
	  Original commit message from CVS:
	  2006-02-01  Philippe Kalaf <burger at speedy dot org>
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
	  setting queue_delay to zero. Also avoid thread being started if
	  queue_delay is zero.

2006-02-01 14:51:29 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
	  Original commit message from CVS:
	  * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
	  Make test work again by connecting fakesinks to each decoded pad,
	  which makes the pipeline wait until each fakesink has a buffer
	  queued before going to PAUSED state. At that point we know the
	  decodebin pads are negotiated.

2006-02-01 11:59:47 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/: Pass unhandled queries to the parent class's query function.
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
	  (gst_cdda_base_src_handle_event):
	  * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
	  Pass unhandled queries to the parent class's query function.

2006-02-01 11:56:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
	  (gst_ogg_pad_src_query):
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
	  * ext/theora/theoradec.c: (theora_dec_src_query),
	  (theora_dec_sink_query):
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
	  (vorbis_dec_sink_query):
	  * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
	  (gst_vorbisenc_sink_query):
	  * gst/adder/gstadder.c: (gst_adder_query):
	  Pass unhandled queries upstream instead of just
	  dropping them (#326447). Also, fix supported
	  query types list for some elements.

2006-02-01 09:58:15 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (au_type_find),
	  (paris_type_find), (ilbc_type_find), (plugin_init):
	  Fix typefinding for audio/x-au, audio/x-paris and
	  audio/iLBC-sh. We cannot use the START_WITH macros
	  here, because there can only be one typefind factory
	  with the same name (caps), so the second one would
	  replace the first one and the first one would never
	  be called when doing typefinding (see #161712).

2006-01-31 19:25:10 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
	  (vorbis_handle_header_packet), (vorbis_dec_push),
	  (vorbis_handle_data_packet):
	  Use scale_int when we can, add some more scaling.
	  Check packettype before parsing it.

2006-01-31 17:44:35 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/theoradec.c: Call right _scale functions.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (_theora_granule_time),
	  (theora_dec_src_convert), (theora_dec_sink_convert):
	  Call right _scale functions.
	  Use parameter instead of some other random value.

2006-01-31 17:27:00 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/theora/theoradec.c: Use higher precision timestamps calculation.
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (_theora_granule_frame),
	  (_theora_granule_time), (_inc_granulepos),
	  (theora_dec_src_convert), (theora_dec_sink_convert),
	  (theora_handle_type_packet), (theora_handle_data_packet),
	  (theora_dec_chain):
	  Use higher precision timestamps calculation.
	  Convert some other conversions to _scale.

2006-01-31 17:19:09 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/: initialize gst_controller before using
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  (gst_audio_test_src_create_sine_table), (plugin_init):
	  * gst/volume/gstvolume.c: (plugin_init):
	  initialize gst_controller before using

2006-01-31 16:26:57 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
	  Original commit message from CVS:
	  * tests/check/pipelines/theoraenc.c:
	  * tests/check/pipelines/vorbisenc.c:
	  Define constant using G_GINT64_CONSTANT to avoid errors when
	  passing it around - otherwise it gets truncated to 32 bits.
	  Fixes failing tests.

2006-01-31 15:36:13 +0000  Andy Wingo <wingo@pobox.com>

	  sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
	  Original commit message from CVS:
	  2006-01-31  Andy Wingo  <wingo@pobox.com>
	  * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
	  caps being set doesn't have a framerate value. Basically a stopgap
	  measure.
	  * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
	  technically correct enough to put into core though.
	  (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
	  DURATION. Fixes theoraenc ! oggmux.
	  * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
	  fraction, not double.

2006-01-31 12:23:35 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  update with latest files
	  Original commit message from CVS:
	  update with latest files

2006-01-30 23:42:54 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/vs7: add vs7 project files created by Sergey Scobich
	  Original commit message from CVS:
	  * win32/vs7:
	  add vs7 project files created by Sergey Scobich

2006-01-30 22:18:53 +0000  Sébastien Moutte <sebastien@moutte.net>

	  win32/vs8: add vs8 project files created by Sergey Scobich
	  Original commit message from CVS:
	  * win32/vs8:
	  add vs8 project files created by Sergey Scobich

2006-01-30 19:22:22 +0000  Andy Wingo <wingo@pobox.com>

	  ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
	  Original commit message from CVS:
	  2006-01-30  Andy Wingo  <wingo@pobox.com>
	  * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
	  timestamp + duration, not just timestamp -- ogg pages should be
	  ordered by stop time. Necessary fix given the change in vorbis
	  timestamps.

2006-01-30 19:21:07 +0000  Andy Wingo <wingo@pobox.com>

	* ChangeLog:
	* ext/theora/gsttheoraenc.h:
	* ext/theora/theoraenc.c:
	* tests/check/pipelines/theoraenc.c:
	  ext/theora/theoraenc.c (theora_enc_sink_setcaps)
	  Original commit message from CVS:
	  2006-01-30  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
	  (gst_theora_enc_init): Pull the granule shift out of the encoder.
	  (granulepos_add): New function, handles the messiness of adjusting
	  granulepos values.
	  (theora_buffer_from_packet):
	  (theora_enc_chain):
	  (theora_enc_sink_event): Use granulepos_add, not +.
	  * tests/check/pipelines/theoraenc.c
	  (check_buffer_granulepos_from_starttime): Just check the frame
	  count, not the actual granulepos -- we can't dictate to the
	  encoder when it should be placing keyframes.

2006-01-30 18:17:19 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
	  SERVICE_NOT_AVAILABLE happens for example when you're trying to
	  play an http:// stream from a server that's not serving

2006-01-30 17:08:11 +0000  Andy Wingo <wingo@pobox.com>

	  tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
	  Original commit message from CVS:
	  2006-01-30  Andy Wingo  <wingo@pobox.com>
	  * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
	  * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
	  remove the UINT64_CONSTANT macro, doesn't appear to be needed or
	  available.

2006-01-30 17:01:54 +0000  Andy Wingo <wingo@pobox.com>

	  ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
	  Original commit message from CVS:
	  2006-01-30  Andy Wingo  <wingo@pobox.com>
	  * ext/theora/gsttheoraenc.h:
	  * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
	  although theoraenc was timestamping correctly. Added handling of
	  streams that start with nonzero timestamps.
	  * tests/check/Makefile.am:
	  * tests/check/pipelines/theoraenc.c: New file, basically does same
	  tests as vorbisenc.
	  * tests/check/pipelines/vorbisenc.c: I claim these bugs.

2006-01-30 16:19:33 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiosink.c:
	  (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
	  (gst_audioringbuffer_pause):
	  Implement pause that does not wait for completion.
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
	  Don't drop buffers when going to PAUSED but perform preroll on
	  remaining samples now that core base class supports this.
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
	  (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
	  (gst_ring_buffer_commit):
	  Pause should not signal waiters.
	  Implement return value of _commit correctly.

2006-01-30 15:01:28 +0000  Andy Wingo <wingo@pobox.com>

	  tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
	  Original commit message from CVS:
	  2006-01-30  Andy Wingo  <wingo@pobox.com>
	  * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
	  * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
	  updated to timestamp from the first sample, not the last.
	  (gst_vorbisenc_buffer_from_header_packet): New function, takes
	  special care of granulepos and timestamp for header packets.
	  (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
	  when the first buffer has a nonzero timestamp.
	  * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
	  (GstVorbisEnc.subgranule_offset): New members. Take care of the
	  case when the first audio buffer we get has a nonzero timestamp.
	  (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
	  properly timestamp vorbis buffers with the time of the first
	  sample, not the last.
	  * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
	  vorbis_granule_time_copy -- now it takes the granule/subgranule
	  offset into account.
	  * tests/check/pipelines/vorbisenc.c: New test for correctness of
	  timestamps, durations, and granulepos on buffers produced by
	  vorbisenc.

2006-01-30 14:42:28 +0000  Eric Jonas <jonas@mit.edu>

	  gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
	  Original commit message from CVS:
	  * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	  (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
	  Patch from Eric Jonas to support conversions to/from UYVY
	  (Fixes: #324626)

2006-01-30 08:11:14 +0000  Julien Moutte <julien@moutte.net>

	  gst/playback/: Implement subtitles.
	  Original commit message from CVS:
	  2006-01-30  Julien MOUTTE  <julien@moutte.net>
	  * gst/playback/gstplaybasebin.c: (group_commit),
	  (queue_overrun),
	  (setup_subtitle), (setup_source), (set_active_source):
	  * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
	  (gen_text_element), (gen_audio_element), (gen_vis_element),
	  (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.

2006-01-29 19:13:39 +0000  Sébastien Moutte <sebastien@moutte.net>

	  gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
	  Original commit message from CVS:
	  * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
	  * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
	  use of gst_guint64_to_gdouble to be compliant with vs6
	  * gst/playback/gstdecodebin.c: (try_to_link_1)
	  * gst/videorate/videorate.c: (gst_video_rate_blank_data)
	  use of G_GINT64_CONSTANT for int64 constants
	  * win32/common/libgstinterfaces.def:
	  export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
	  * win32/vs6:
	  update and add new project files

2006-01-29 18:21:12 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  add a win32-update rule like in core, and copy over enumtypes files
	  Original commit message from CVS:
	  * Makefile.am:
	  * win32/MANIFEST:
	  * win32/common/interfaces-enumtypes.c:
	  (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
	  (gst_mixer_track_flags_get_type),
	  (gst_tuner_channel_flags_get_type):
	  * win32/common/interfaces-enumtypes.h:
	  * win32/common/multichannel-enumtypes.c:
	  (gst_audio_channel_position_get_type):
	  * win32/common/multichannel-enumtypes.h:
	  add a win32-update rule like in core, and copy over enumtypes files

2006-01-29 18:07:51 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  generate win32/common/config.h
	  Original commit message from CVS:
	  generate win32/common/config.h

2006-01-29 18:05:42 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  win32/: add config files just like in core
	  Original commit message from CVS:
	  * win32/MANIFEST:
	  * win32/common/config.h:
	  * win32/common/config.h.in:
	  add config files just like in core

2006-01-28 18:22:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/alsa/: Update all error messages.  All of them should either use the default translated message, or actually prov...
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
	  (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
	  (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
	  * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
	  (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
	  (gst_alsasrc_unprepare), (gst_alsasrc_read):
	  Update all error messages.  All of them should either use
	  the default translated message, or actually provide a
	  translatable string.
	  Make the string for channel count problems meaningful.

2006-01-28 18:19:18 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
	  Make gcc-4.1 happy (part of #327357).

2006-01-28 16:35:47 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
	  Original commit message from CVS:
	  * sys/v4l/v4l_calls.c: (gst_v4l_open):
	  check for and throw RESOURCE_BUSY

2006-01-28 02:13:14 +0000  David Schleef <ds@schleef.org>

	  gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
	  Original commit message from CVS:
	  * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
	  checked in this change -- it requires liboil features not
	  in 0.3.6.  Revert parts.

2006-01-27 23:40:19 +0000  David Schleef <ds@schleef.org>

	  update liboil requirement to 0.3.6
	  Original commit message from CVS:
	  * REQUIREMENTS:
	  * configure.ac: update liboil requirement to 0.3.6
	  * gst/videoscale/Makefile.am:
	  * gst/videoscale/vs_scanline.c: liboilify

2006-01-27 17:00:09 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (get_buffer):
	  When pad_alloc returns a GstFlowReturn other
	  than GST_FLOW_OK, make sure it is passed upstream.

2006-01-27 01:36:01 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/alsa/gstalsasink.c: Free the device name string.
	  Original commit message from CVS:
	  * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
	  (gst_alsasink_class_init):
	  Free the device name string.
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
	  (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
	  (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
	  Don't remove a pad from the collectpads structure until it
	  is released - it's a request pad, and may receive data again
	  if the element gets moved back to PLAYING state.
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
	  Ensure we turn on double buffering on the Xv port, and
	  set the colour key to something dark and mysterious that
	  isn't black.

2006-01-27 01:06:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/: - a library should not call setlocale. see Libraries node in gettext manual
	  Original commit message from CVS:
	  * ext/alsa/gstalsaplugin.c: (plugin_init):
	  * ext/cdparanoia/gstcdparanoiasrc.c:
	  (gst_cd_paranoia_src_base_init), (plugin_init):
	  * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
	  - a library should not call setlocale. see Libraries node in
	  gettext manual
	  - make sure all plugins that use translation do bindtextdomain
	  to point to the localedir
	  * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
	  (setup_sinks), (plugin_init):
	  all this, and check for NULL when creating sinks

2006-01-26 23:21:31 +0000  Julien Moutte <julien@moutte.net>

	  gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
	  Original commit message from CVS:
	  2006-01-27  Julien MOUTTE  <julien@moutte.net>
	  * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
	  (plugin_init): Make typefinding of subtitles work again.

2006-01-26 20:40:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
	  Original commit message from CVS:
	  * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
	  (mp3_type_frame_length_from_header), (mp3_type_find),
	  (wavpack_type_find), (m4a_type_find), (ircam_type_find),
	  (plugin_init):
	  Backport a bunch of typefinding fixes from the 0.8 branch.
	  Also, improve wavpack typefinding: if we can't peek the
	  entire wavpack block, try to parse the bits we can get and
	  see if we find what we're looking for in those.

2006-01-26 19:17:38 +0000  Julien Moutte <julien@moutte.net>

	  sys/: Handle some more cases of pixel aspect ratio.
	  Original commit message from CVS:
	  2006-01-26  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c:
	  (gst_ximagesink_calculate_pixel_aspect_ratio):
	  * sys/xvimage/xvimagesink.c:
	  (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
	  more cases of pixel aspect ratio.

2006-01-26 13:09:24 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (pad_probe):
	  Also consider the flush-start and tag events as unblockers
	  for the pad probes.

2006-01-26 12:32:58 +0000  Julien Moutte <julien@moutte.net>

	  gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
	  Original commit message from CVS:
	  2006-01-26  Julien MOUTTE  <julien@moutte.net>
	  * gst/playback/gstplaybin.c: (gst_play_bin_init),
	  (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
	  (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
	  On the fly visualisation switch, works disabling, enabling as
	  well but it won't be able to enable vis in a playbin that was
	  created with no visualisation.

2006-01-25 10:50:32 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
	  Undo previous commit, it breaks resume after pause.

2006-01-25 09:27:01 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
	  (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
	  Improve debugging.
	  Post error when caps cannot be parsed.
	  Resync on discontinuity in the stream.
	  Clip samples to segment boundaries.
	  return WRONG_STATE sooner when we are flushing.
	  * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
	  (gst_base_audio_src_get_time), (gst_base_audio_src_create):
	  Make audiosrc operate in TIME.
	  Set TIMESTAMP and DURATION on buffers.

2006-01-24 21:55:21 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/examples/seek/seek.c: Output tag messages as well.
	  Original commit message from CVS:
	  * tests/examples/seek/seek.c: (main):
	  Output tag messages as well.

2006-01-23 15:05:24 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
	  (free_pad_probes), (remove_fakesink), (pad_probe),
	  (close_pad_link), (gst_decode_bin_change_state):
	  Replace GstPadBlockCallback with pad probes that detect
	  first buffer AND eos before removing fakesink.
	  Fixes hang with demuxers doing EOS while pre-rolling.
	  Solves #328279

2006-01-23 10:10:36 +0000  Jens Granseuer <jensgr@gmx.net>

	  GCC 2.95 fixes (#328263).
	  Original commit message from CVS:
	  2006-01-23  Andy Wingo  <wingo@pobox.com>
	  * ext/alsa/gstalsasink.c:
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_setcaps),
	  (gst_base_rtp_depayload_add_to_queue),
	  (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
	  Patch by: Jens Granseuer <jensgr at gmx dot net>

2006-01-22 17:24:02 +0000  Julien Moutte <julien@moutte.net>

	  sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
	  Original commit message from CVS:
	  2006-01-22  Julien MOUTTE  <julien@moutte.net>
	  * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
	  (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
	  frames. We might get a frame destroyed after changing state to
	  NULL, adding a safety check on xcontext.

2006-01-22 14:50:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
	  Original commit message from CVS:
	  * gst-libs/gst/interfaces/xoverlay.c:
	  Fix prepare-xwindow-id code example in the docs - we need to
	  ignore all messages that aren't element messages as well.

2006-01-21 22:40:03 +0000  Julien Moutte <julien@moutte.net>

	  sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
	  Original commit message from CVS:
	  2006-01-21  Julien MOUTTE  <julien@moutte.net>
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
	  I think one day i'll completely undestand how caps negotiation
	  is supposed to work. This refactoring handles buffer_alloc
	  called with caps we can't handle. We definitely don't want a
	  set_caps with those caps, so we define and allocate a buffer
	  we would like to receive.

2006-01-20 19:10:26 +0000  Christian Schaller <uraeus@gnome.org>

	* autogen.sh:
	* common:
	  up automake requirement to 1.7
	  Original commit message from CVS:
	  up automake requirement to 1.7

2006-01-19 10:59:51 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Free iterator when done.
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_source):
	  Free iterator when done.

2006-01-17 11:43:49 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_render):
	  Fix playback of non-synchronised streams by assuming a rate
	  of 1.0 instead of a random one.
	  Makes this work again:
	  gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
	  endianness=(int)4321, signed=(boolean)true, width=(int)16,
	  depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
	  audioresample ! alsasink

2006-01-16 21:01:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.10.2 ===

2006-01-16 20:59:32 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	  releasing 0.10.2
	  Original commit message from CVS:
	  releasing 0.10.2

2006-01-16 16:38:15 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/: Comment out broken code that connects to the state-changed signal.
	  Original commit message from CVS:
	  * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_set_property):
	  Comment out broken code that connects to the state-changed signal.
	  At this point, changing current stream selection is broken, but
	  stuff like gst-launch playbin current-audio=1 works and filters
	  to the chosen stream.

2006-01-16 15:31:14 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
	  Original commit message from CVS:
	  * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
	  Fix #327216 (null dereference in vorbisdec)

2006-01-16 15:19:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
	  Original commit message from CVS:
	  * ext/theora/theoradec.c: (theora_handle_comment_packet):
	  Post taglist actually on bus instead of just freeing it
	  (fixes #327114 and totem bug #327080).
	  * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
	  Use gst_element_found_tags_for_pad(), so that the tags
	  are sent downstream as an event as well.

2006-01-15 10:06:40 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  sys/: move all regularly occurring messages to GST_LOG level add some more object logs
	  Original commit message from CVS:
	  * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
	  (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
	  (gst_ximagesink_buffer_alloc):
	  * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
	  (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
	  (gst_xvimagesink_buffer_alloc):
	  move all regularly occurring messages to GST_LOG level
	  add some more object logs

2006-01-14 22:59:49 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  prerelease
	  Original commit message from CVS:
	  prerelease

2006-01-14 20:46:25 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  ext/ogg/gstoggmux.c: fix a silly segfault
	  Original commit message from CVS:
	  2006-01-14  Thomas Vander Stichele  <thomas at apestaart dot org>
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
	  fix a silly segfault

2006-01-14 12:52:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add docs for mixerutils stuff.
	  Original commit message from CVS:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * gst-libs/gst/audio/mixerutils.c:
	  * gst-libs/gst/audio/mixerutils.h:
	  Add docs for mixerutils stuff.

2006-01-13 17:17:07 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (setup_source):
	  Fix playback for sources that emit raw audio or
	  raw video streams (e.g.: cd audio sources) (#325984).

2006-01-13 16:45:50 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst-libs/gst/audio/mixerutils.c: actually save the element we create
	  Original commit message from CVS:
	  * gst-libs/gst/audio/mixerutils.c:
	  (gst_audio_mixer_filter_do_filter):
	  actually save the element we create

2006-01-13 16:17:50 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  remove version suffix
	  Original commit message from CVS:
	  remove version suffix

2006-01-12 14:56:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_handle_track_seek):
	  No need to post a tag message on the bus when seeking
	  within the same track, only post it when the current
	  track changes.

2006-01-11 18:30:25 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (group_destroy),
	  (probe_triggered), (new_decoded_pad), (mute_group_type),
	  (set_active_source):
	  * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
	  * gst/playback/gststreamselector.c:
	  (gst_stream_selector_base_init),
	  (gst_stream_selector_set_property),
	  (gst_stream_selector_request_new_pad):
	  Reenable stream selection. These mechanisms need a complete overhaul
	  in the face of 0.8->0.10 changes though.

2006-01-11 18:03:24 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c:
	  Change the pad template to src_%d to match the pads that
	  are created from it. decodebin needs this information in order
	  to decide that oggdemux is capable of producing multiple pads
	  (and hence needs queues inserted).
	  * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
	  (gst_ogg_mux_collected):
	  Make debug output more useful by using GST_PTR_FORMAT.

2006-01-11 17:38:35 +0000  Christian Schaller <uraeus@gnome.org>

	* gst-plugins-base.spec.in:
	  update spec.in file
	  Original commit message from CVS:
	  update spec.in file

2006-01-11 15:11:20 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
	  Original commit message from CVS:
	  Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
	  Set depth and width for alaw/mulaw (fixes #326601).

2006-01-10 23:58:36 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  tests/icles/Makefile.am: don't build the tests if we don't have the libs
	  Original commit message from CVS:
	  * tests/icles/Makefile.am:
	  don't build the tests if we don't have the libs

2006-01-10 18:06:56 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
	  Original commit message from CVS:
	  * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
	  (gst_cd_paranoia_paranoia_callback):
	  Don't try to free NULL pointers.

2006-01-10 15:47:48 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst/audiorate/gstaudiorate.c: Add debugging category.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
	  (gst_audio_rate_change_state), (plugin_init):
	  Add debugging category.
	  Fix type issues.
	  Add case for incoming buffers without valid offset/offset_end.

2006-01-10 12:25:59 +0000  Michael Smith <msmith@xiph.org>

	  gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
	  Don't leak GCond in audio sources.

2006-01-10 11:49:28 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
	  Original commit message from CVS:
	  * gst/playback/gstplaybin.c: (gen_audio_element):
	  Don't leak an autoaudiosink/alsasink when we generate
	  a new audio element. (old code, I guess)

2006-01-10 11:04:21 +0000  Michael Smith <msmith@xiph.org>

	  gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
	  Original commit message from CVS:
	  * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
	  Support float audio in audiorate.
	  Use width rather than depth for selecting sample width.

2006-01-10 10:06:53 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
	  Original commit message from CVS:
	  * gst/videotestsrc/videotestsrc.h:
	  Use GLib types here (that way we don't have to include the
	  generated _stdint.h header, which makes life easier for win32
	  folks that don't use autotools for the build) (#325990, patch
	  by: Sergey Scobich).

2006-01-10 09:38:44 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
	  (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
	  (gst_ring_buffer_pause), (wait_segment):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Name (private) union, makes Forte compiler happy (this time
	  for real) (#324900).

2006-01-09 10:52:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/Makefile.am:
	  Link against libgstinterfaces, needed for mixer
	  and property probe stuff.

2006-01-09 10:46:52 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst-libs/gst/Makefile.am:
	  Original commit message from CVS:
	  * gst-libs/gst/Makefile.am:

2006-01-09 09:38:34 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
	  Original commit message from CVS:
	  * gst-libs/gst/audio/Makefile.am:
	  * gst-libs/gst/audio/mixerutils.c:
	  (gst_audio_mixer_filter_do_filter),
	  (gst_audio_mixer_filter_check_element),
	  (gst_audio_mixer_filter_probe_feature),
	  (element_factory_rank_compare_func),
	  (gst_audio_default_registry_mixer_filter):
	  * gst-libs/gst/audio/mixerutils.h:
	  Add gst_audio_default_registry_mixer_filter() utility
	  function.

2006-01-03 17:33:16 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioresample/resample.h: As before, but for o_buf
	  Original commit message from CVS:
	  * gst/audioresample/resample.h:
	  As before, but for o_buf

2006-01-03 17:27:13 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
	  Original commit message from CVS:
	  * gst/audioresample/resample.h:
	  Declare struct _ResampleState.buffer as unsigned char *, not void *,
	  since we do arithmetic on it.

2006-01-02 23:37:38 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
	  (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
	  (gst_ring_buffer_pause), (wait_segment):
	  * gst-libs/gst/audio/gstringbuffer.h:
	  Sun's Forte compiler doesn't seem to like anonymous structs,
	  so use same setup as in GstBaseSrc (fixes #324900).

2005-12-30 14:54:06 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  move old example to tests/examples/volume/volune.c
	  Original commit message from CVS:
	  * configure.ac:
	  * gst/volume/Makefile.am:
	  * gst/volume/demo.c:
	  move old example to tests/examples/volume/volune.c
	  * tests/examples/Makefile.am:
	  * tests/examples/seek/seek.c: (main):
	  change window-close event from "delete-event" to "destroy"
	  * tests/examples/volume/Makefile.am:
	  * tests/examples/volume/volume.c: (value_changed_callback),
	  (setup_gui), (message_received), (eos_message_received), (main):
	  fix event handling and bus usage

2005-12-29 20:37:23 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
	  (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
	  (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
	  (gst_audio_test_src_create_square),
	  (gst_audio_test_src_create_saw),
	  (gst_audio_test_src_create_triangle),
	  (gst_audio_test_src_create_silence),
	  (gst_audio_test_src_create_white_noise),
	  (gst_audio_test_src_create_pink_noise),
	  (gst_audio_test_src_init_sine_table),
	  (gst_audio_test_src_create_sine_table),
	  (gst_audio_test_src_change_wave),
	  (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
	  (gst_audio_test_src_create), (gst_audio_test_src_set_property):
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  update to basesrc changes, implement segmented seeking and eos handling,
	  add a 'sine-tab' waveform for performance critical playback

2005-12-29 16:17:55 +0000  Tim-Philipp Müller <tim@centricular.net>

	  po/POTFILES.in: ... and this time the other modified file that I missed last time.
	  Original commit message from CVS:
	  * po/POTFILES.in:
	  ... and this time the other modified file that I missed last time.

2005-12-29 14:31:49 +0000  Michael Smith <msmith@xiph.org>

	  gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
	  Original commit message from CVS:
	  * gst/playback/gstdecodebin.c: (new_pad):
	  Fix non-C89 variable declaration not at the start of a block. Should
	  help some compilers.

2005-12-29 12:43:22 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  And now fix 'make distcheck' (builddir != srcdir)

2005-12-29 12:22:24 +0000  Tim-Philipp Müller <tim@centricular.net>

	  New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
	  Original commit message from CVS:
	  * configure.ac:
	  * ext/cdparanoia/Makefile.am:
	  * ext/cdparanoia/gstcdparanoia.c:
	  * ext/cdparanoia/gstcdparanoia.h:
	  * ext/cdparanoia/gstcdparanoiasrc.c:
	  (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
	  (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
	  (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
	  (gst_cd_paranoia_paranoia_callback),
	  (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
	  (gst_cd_paranoia_src_set_property),
	  (gst_cd_paranoia_src_get_property), (plugin_init):
	  * ext/cdparanoia/gstcdparanoiasrc.h:
	  New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
	  plugin again (there are still fixes required to playbin to make
	  cdda:// uris work there).

2005-12-29 12:13:57 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/Makefile.am: Fix test case compilation.
	  Original commit message from CVS:
	  * tests/check/Makefile.am:
	  Fix test case compilation.

2005-12-29 11:49:11 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
	  Original commit message from CVS:
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_update_duration),
	  (gst_cdda_base_src_calculate_cddb_id):
	  An integer is not a string. Fix access to uninitialised variable.
	  * tests/check/Makefile.am:
	  Add cddabasesrc unit test; also actually enable the vorbis test.
	  * tests/check/generic/states.c:
	  Blacklist new cd audio elements as well.
	  * tests/check/libs/cddabasesrc.c:
	  Unit test for GstCddaBaseSrc (discid calculation mostly).

2005-12-28 18:19:25 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
	  Original commit message from CVS:
	  * docs/libs/Makefile.am:
	  * docs/libs/gst-plugins-base-libs-docs.sgml:
	  * docs/libs/gst-plugins-base-libs-sections.txt:
	  * docs/libs/gst-plugins-base-libs.types:
	  Add docs for libgstcdda/GstCddaBaseSrc.
	  * gst-libs/gst/interfaces/mixertrack.h:
	  Do one struct member per line with a semicolon at the end, that way
	  even gtk-doc might parse it without complaining.

2005-12-28 18:06:50 +0000  Tim-Philipp Müller <tim@centricular.net>

	  Add new libgstcdda with GstCddaBaseSrc class.
	  Original commit message from CVS:
	  * configure.ac:
	  * gst-libs/gst/Makefile.am:
	  * gst-libs/gst/cdda/Makefile.am:
	  * gst-libs/gst/cdda/base64.c:
	  * gst-libs/gst/cdda/base64.h:
	  * gst-libs/gst/cdda/gstcddabasesrc.c:
	  (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
	  (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
	  (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
	  (gst_cdda_base_src_get_property),
	  (gst_cdda_base_src_get_track_from_sector),
	  (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
	  (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
	  (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
	  (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
	  (gst_cdda_base_src_uri_get_protocols),
	  (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
	  (gst_cdda_base_src_uri_handler_init),
	  (gst_cdda_base_src_setup_interfaces),
	  (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
	  (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
	  (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
	  (gst_cdda_base_src_add_tags),
	  (gst_cdda_base_src_add_index_associations),
	  (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
	  (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
	  (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
	  (gst_cdda_base_src_create):
	  * gst-libs/gst/cdda/gstcddabasesrc.h:
	  * gst-libs/gst/cdda/sha1.c:
	  * gst-libs/gst/cdda/sha1.h:
	  Add new libgstcdda with GstCddaBaseSrc class.

2005-12-28 14:59:41 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
	  Original commit message from CVS:
	  * ext/gnomevfs/gstgnomevfssink.h:
	  Use GstBaseSinkClass as parent_class member for class struct, not
	  GstBaseSink.

2005-12-27 22:29:43 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  (gst_video_test_src_class_init), (gst_video_test_src_start):
	  Add start method to reset running time and number of frames sent
	  when starting up (fixes #324696; patch by: Michal Benes).

2005-12-27 21:58:28 +0000  Tim-Philipp Müller <tim@centricular.net>

	  docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
	  Original commit message from CVS:
	  * docs/plugins/Makefile.am:
	  * docs/plugins/gst-plugins-base-plugins-docs.sgml:
	  * docs/plugins/gst-plugins-base-plugins-sections.txt:
	  * docs/plugins/gst-plugins-base-plugins.args:
	  * docs/plugins/gst-plugins-base-plugins.hierarchy:
	  * docs/plugins/gst-plugins-base-plugins.signals:
	  Add docs stuff for gnomevfssrc and gnomevfssink.
	  * ext/gnomevfs/gstgnomevfssrc.c:
	  Fix example pipeline in gtk-doc blurb.

2005-12-27 21:42:23 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
	  Original commit message from CVS:
	  * ext/gnomevfs/Makefile.am:
	  * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
	  (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
	  (gst_gnome_vfs_handle_get_type), (plugin_init):
	  * ext/gnomevfs/gstgnomevfs.h:
	  * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
	  (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
	  (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
	  (gst_gnome_vfs_sink_set_property),
	  (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
	  (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
	  (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
	  (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
	  (gst_gnome_vfs_sink_uri_get_type),
	  (gst_gnome_vfs_sink_uri_get_protocols),
	  (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
	  (gst_gnome_vfs_sink_uri_handler_init):
	  * ext/gnomevfs/gstgnomevfssink.h:
	  Port gnomevfssink; add gtk-doc blurb.
	  * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
	  (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
	  (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
	  (gst_gnome_vfs_src_uri_get_type),
	  (gst_gnome_vfs_src_uri_get_protocols),
	  (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
	  (gst_gnome_vfs_src_uri_handler_init),
	  (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
	  (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
	  (gst_gnome_vfs_src_send_additional_headers_callback),
	  (gst_gnome_vfs_src_received_headers_callback),
	  (gst_gnome_vfs_src_push_callbacks),
	  (gst_gnome_vfs_src_pop_callbacks),
	  (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
	  (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
	  (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
	  * ext/gnomevfs/gstgnomevfssrc.h:
	  s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
	  file; add gtk-doc blurb with example pipelines.

2005-12-23 18:16:22 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.10.1 ===

2005-12-23 18:08:39 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.signals:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	  releasing 0.10.1
	  Original commit message from CVS:
	  releasing 0.10.1

2005-12-21 20:59:52 +0000  Edgard Lima <edgard.lima@indt.org.br>

	* ChangeLog:
	* gst/typefind/gsttypefindfunctions.c:
	  iLBC30 and iLBC20 added to typefind.
	  Original commit message from CVS:
	  iLBC30 and iLBC20 added to typefind.

2005-12-20 15:57:06 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* configure.ac:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/hu.po:
	* po/it.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  prereleasing
	  Original commit message from CVS:
	  prereleasing

2005-12-20 12:24:29 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	  stop making fun of older compilers
	  Original commit message from CVS:
	  stop making fun of older compilers

2005-12-20 12:00:26 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	  gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
	  Original commit message from CVS:
	  * gst-libs/gst/audio/gstbaseaudiosink.c:
	  (gst_base_audio_sink_class_init):
	  * gst-libs/gst/audio/gstbaseaudiosrc.c:
	  (gst_base_audio_src_class_init):
	  update strings, values are in microseconds
	  change the default sink buffer time to something that is smaller
	  (to help software volume mixing have a slightly lower delay) but
	  still be acceptable on Wim's laptop

2005-12-20 10:13:05 +0000  Edward Hervey <bilboed@bilboed.com>

	  gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
	  Original commit message from CVS:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
	  Made a quack, forgot to add DUCK to the riff video template.

2005-12-19 15:00:38 +0000  Edward Hervey <bilboed@bilboed.com>

	  ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
	  Original commit message from CVS:
	  * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
	  (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
	  (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
	  (gst_ogm_parse_chain):
	  Make sure pads are initialized correctly.
	  * gst-libs/gst/riff/riff-ids.h:
	  * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
	  (gst_riff_create_video_template_caps):
	  Add a whole bunch of FOURCC <=> MimeType.
	  Extend the riff video pad template to support the newly added fourcc.

2005-12-18 15:04:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>

	  ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
	  (gst_ogg_demux_activate_chain):
	  Extra debug output when activating/deactivating chains.
	  * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
	  (is_demuxer_element), (try_to_link_1), (remove_element_chain),
	  (unlinked):
	  Remove a queue from our list when it becomes unlinked.
	  Don't add queues to elements in class 'Demux' if they
	  can only produce one pad

2005-12-18 10:54:33 +0000  Julien Moutte <julien@moutte.net>

	  gst-libs/gst/video/gstvideosink.c: Add a debug category.
	  Original commit message from CVS:
	  2005-12-18  Julien MOUTTE  <julien@moutte.net>
	  * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
	  (gst_video_sink_get_type): Add a debug category.

2005-12-18 00:56:07 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
	  Original commit message from CVS:
	  2005-12-17  Philippe Khalaf  <burger@speedy.org>
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
	  Handle downstream newsegment by sending our own newsegment before the
	  next buffer to be released. (#323900)

2005-12-18 00:41:10 +0000  Philippe Kalaf <philippe.kalaf@collabora.co.uk>

	  gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
	  Original commit message from CVS:
	  2005-12-17  Philippe Khalaf  <burger@speedy.org>
	  * gst-libs/gst/rtp/gstbasertpdepayload.c:
	  (gst_base_rtp_depayload_set_gst_timestamp):
	  add queue delay to new segment as well (as opposed to just the first
	  buffer). (bug #322347)

2005-12-16 22:00:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  ext/libvisual/visual.c: change some char* into char[]
	  Original commit message from CVS:
	  * ext/libvisual/visual.c: (make_valid_name):
	  change some char* into char[]
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
	  (gst_audio_test_src_create):
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  prepare to handle EOS and SEGMENT_DONE

2005-12-16 12:32:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  tests/check/generic/states.c: Blacklist cdparanoia element in state test.
	  Original commit message from CVS:
	  * tests/check/generic/states.c: (GST_START_TEST):
	  Blacklist cdparanoia element in state test.

2005-12-16 11:25:51 +0000  Benjamin Pineau <ben.pineau@gmail.com>

	  gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
	  Original commit message from CVS:
	  * gst/tcp/gsttcp.c:
	  * gst/tcp/gsttcpclientsink.c:
	  * gst/tcp/gsttcpserversink.c:
	  * gst/tcp/gsttcpserversrc.c:
	  Add <string.h> includes for memset and FD_ZERO (fixes #323878;
	  patch by: Benjamin Pineau).

2005-12-15 14:43:38 +0000  Michael Smith <msmith@xiph.org>

	  gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
	  Original commit message from CVS:
	  * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
	  (gst_video_rate_chain):
	  Fix timestamping for videorate when the first buffer it sees has a
	  non-zero timestamp. Fix some misleading debug output.

2005-12-15 10:30:14 +0000  Michael Smith <msmith@xiph.org>

	  gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
	  Original commit message from CVS:
	  * gst/audioresample/gstaudioresample.c:
	  Don't leak all input buffers to audioresample.

2005-12-15 10:15:10 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
	  Original commit message from CVS:
	  * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
	  Don't operate on empty text buffers. Strip newlines and
	  tabs only from the end of the text, but leave them intact
	  in the middle. Fix typo in gtk-doc description.

2005-12-15 09:48:19 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c:
	  * gst/playback/gstplaybin.c: (handoff):
	  Make sure the video frame buffer we return to apps via the
	  "frame" property always has caps set on it. Modify
	  _gst_gvalue_set_object() macro to handle NULL objects
	  gracefully too.

2005-12-14 20:42:11 +0000  Stefan Kost <ensonic@users.sourceforge.net>

	  gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
	  Original commit message from CVS:
	  * gst/audiotestsrc/gstaudiotestsrc.c:
	  (gst_audio_test_src_class_init), (gst_audio_test_src_init),
	  (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
	  (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
	  (gst_audio_test_src_create):
	  * gst/audiotestsrc/gstaudiotestsrc.h:
	  Adjust to some recent api changes and add wtays new cool seeking
	  capabillities

2005-12-14 17:58:48 +0000  Tim-Philipp Müller <tim@centricular.net>

	  ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
	  Original commit message from CVS:
	  * ext/alsa/Makefile.am:
	  * ext/alsa/gstalsadeviceprobe.c:
	  * ext/alsa/gstalsadeviceprobe.h:
	  Helper functions to add device probing via the GstPropertyProbe
	  interface to a class.
	  * ext/alsa/gstalsamixer.h:
	  Comment out GST_ALSA_MIXER, it returns a struct that's not
	  used.
	  * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
	  Add some debug info.
	  * ext/alsa/gstalsamixerelement.c:
	  (gst_alsa_mixer_element_interface_supported),
	  (gst_implements_interface_init),
	  (gst_alsa_mixer_element_init_interfaces),
	  (gst_alsa_mixer_element_class_init),
	  (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
	  (gst_alsa_mixer_element_set_property),
	  (gst_alsa_mixer_element_get_property),
	  (gst_alsa_mixer_element_change_state):
	  * ext/alsa/gstalsamixerelement.h:
	  Add 'device' and 'device-name' properties. Add GstPropertyProbe
	  for device handling (gnome-volume-control will need that).

2005-12-12 20:31:24 +0000  Christian Schaller <uraeus@gnome.org>

	* ChangeLog:
	* ext/Makefile.am:
	* gst-plugins-base.spec.in:
	  updates to activate cdparanoia plugin
	  Original commit message from CVS:
	  updates to activate cdparanoia plugin

2005-12-12 19:13:09 +0000  Michael Smith <msmith@xiph.org>

	  ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
	  Original commit message from CVS:
	  * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
	  Use the correct function to free list of typefind factories.

2005-12-12 15:09:55 +0000  Wim Taymans <wim.taymans@gmail.com>

	  gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
	  Original commit message from CVS:
	  * gst/videotestsrc/gstvideotestsrc.c:
	  (gst_video_test_src_class_init), (gst_video_test_src_init),
	  (gst_video_test_src_parse_caps), (gst_video_test_src_query),
	  (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
	  (gst_video_test_src_create):
	  * gst/videotestsrc/gstvideotestsrc.h:
	  Implement seeking in videotestsrc.
	  Small cleanups.

2005-12-12 15:06:46 +0000  Wim Taymans <wim.taymans@gmail.com>

	  ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
	  Original commit message from CVS:
	  * ext/cdparanoia/Makefile.am:
	  * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
	  (gst_paranoia_endian_get_type), (_do_init),
	  (cdparanoia_class_init), (cdparanoia_init),
	  (cdparanoia_set_property), (cdparanoia_get_property),
	  (cdparanoia_do_seek), (cdparanoia_is_seekable),
	  (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
	  (cdparanoia_convert), (cdparanoia_get_query_types),
	  (cdparanoia_query), (cdparanoia_set_index),
	  (cdparanoia_uri_set_uri):
	  * ext/cdparanoia/gstcdparanoia.h:
	  Partially ported cdparanoia now that basesrc can support a
	  plugin like this..

2005-12-12 13:03:50 +0000  Wim Taymans <wim.taymans@gmail.com>

	  tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
	  Original commit message from CVS:
	  * tests/examples/seek/scrubby.c: (main):
	  Set higher priority for bus events so they don't get reordered with
	  gtk gui events.
	  * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
	  (flush_toggle_cb), (main):
	  Added checkbox do disable flushing seeks.
	  Disable scrubbing when doing non flushing seeks.

2005-12-12 09:52:37 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
	  Original commit message from CVS:
	  * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
	  (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
	  (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
	  (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
	  Implement some sort of event handling that doesn't rely on
	  g_return_if_fail; make sure we always push the last chunk of an
	  .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
	  state change function; remove some old cruft. Seeking is still
	  rather unlikely to work though.
	  * tools/.cvsignore:
	  Ignore more.

2005-12-11 22:26:08 +0000  Julien Moutte <julien@moutte.net>

	  sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
	  Original commit message from CVS:
	  2005-12-11  Julien MOUTTE  <julien@moutte.net>
	  * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
	  Fixed a leak of the current image reference when cleaning up.
	  Thanks to Arwed von Merkatz (alley_cat) for pointing it out.

2005-12-09 10:23:42 +0000  Michael Smith <msmith@xiph.org>

	  tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
	  Original commit message from CVS:
	  * tools/Makefile.am:
	  * tools/gst-launch-ext-m.m:
	  Remove gst-launch-ext. It doesn't work, and is no longer
	  particularly useful.

2005-12-08 18:53:57 +0000  Luca Ognibene <luogni@tin.it>

	  ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
	  Original commit message from CVS:
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
	  don't pass random values to ogmparse convert function.
	  Make seeking possible in the exile1.ogm file.

2005-12-07 18:51:35 +0000  Tim-Philipp Müller <tim@centricular.net>

	  gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
	  Original commit message from CVS:
	  * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
	  * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
	  Work around refcount problem with g_value_set_object() that occur
	  if the core has been compiled against GLib-2.6 (g_value_set_object()
	  will only g_object_ref() the element, but the caller will
	  gst_object_unref() it and bad things will happen due to the way
	  GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
	  totem for people on FC4 using Thomas's 0.10 RPMs.

2005-12-07 11:34:37 +0000  Edward Hervey <bilboed@bilboed.com>

	  Time to welcome ogm to 0.10 :)
	  Original commit message from CVS:
	  Time to welcome ogm to 0.10 :)
	  * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
	  (gst_ogg_pad_typefind):
	  Oggdemux can now properly typefind elements with dynamic pads.
	  * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
	  Properly set caps on src pad, and set caps on outgoing buffers.

2005-12-06 19:42:02 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* ext/alsa/gstalsamixer.h:
	* ext/alsa/gstalsamixerelement.h:
	* ext/alsa/gstalsamixeroptions.h:
	* ext/alsa/gstalsamixertrack.h:
	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasink.h:
	* ext/alsa/gstalsasrc.c:
	* ext/alsa/gstalsasrc.h:
	* ext/cdparanoia/gstcdparanoia.h:
	* ext/gnomevfs/gstgnomevfsuri.h:
	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggmux.c:
	* ext/pango/gsttextoverlay.h:
	* ext/theora/theoradec.c:
	* ext/theora/theoraenc.c:
	* ext/vorbis/vorbisdec.h:
	* ext/vorbis/vorbisenc.c:
	* ext/vorbis/vorbisenc.h:
	* ext/vorbis/vorbisparse.h:
	* gst-libs/gst/audio/gstaudioclock.h:
	* gst-libs/gst/audio/gstaudiosink.c:
	* gst-libs/gst/audio/gstaudiosink.h:
	* gst-libs/gst/audio/gstaudiosrc.c:
	* gst-libs/gst/audio/gstaudiosrc.h:
	* gst-libs/gst/audio/gstbaseaudiosink.c:
	* gst-libs/gst/audio/gstbaseaudiosink.h:
	* gst-libs/gst/audio/gstbaseaudiosrc.c:
	* gst-libs/gst/audio/gstbaseaudiosrc.h:
	* gst-libs/gst/audio/gstringbuffer.h:
	* gst-libs/gst/audio/multichannel.h:
	* gst-libs/gst/floatcast/floatcast.h:
	* gst-libs/gst/interfaces/colorbalance.c:
	* gst-libs/gst/interfaces/colorbalance.h:
	* gst-libs/gst/interfaces/colorbalancechannel.h:
	* gst-libs/gst/interfaces/mixer.h:
	* gst-libs/gst/interfaces/mixeroptions.h:
	* gst-libs/gst/interfaces/mixertrack.h:
	* gst-libs/gst/interfaces/navigation.h:
	* gst-libs/gst/interfaces/propertyprobe.h:
	* gst-libs/gst/interfaces/tuner.h:
	* gst-libs/gst/interfaces/tunerchannel.h:
	* gst-libs/gst/interfaces/tunernorm.h:
	* gst-libs/gst/interfaces/xoverlay.h:
	* gst-libs/gst/netbuffer/gstnetbuffer.h:
	* gst-libs/gst/riff/riff-ids.h:
	* gst-libs/gst/riff/riff-media.h:
	* gst-libs/gst/riff/riff-read.h:
	* gst-libs/gst/rtp/gstbasertpdepayload.h:
	* gst-libs/gst/rtp/gstbasertppayload.c:
	* gst-libs/gst/rtp/gstbasertppayload.h:
	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* gst-libs/gst/rtp/gstrtpbuffer.h:
	* gst-libs/gst/tag/gsttageditingprivate.h:
	* gst-libs/gst/tag/gstvorbistag.c:
	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/video/video.h:
	* gst/adder/gstadder.c:
	* gst/adder/gstadder.h:
	* gst/audioconvert/audioconvert.c:
	* gst/audioconvert/audioconvert.h:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstchannelmix.c:
	* gst/audioconvert/gstchannelmix.h:
	* gst/audiorate/gstaudiorate.c:
	* gst/audioresample/buffer.h:
	* gst/audioresample/functable.h:
	* gst/audioresample/gstaudioresample.c:
	* gst/audioresample/resample.h:
	* gst/ffmpegcolorspace/avcodec.h:
	* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
	* gst/ffmpegcolorspace/gstffmpegcodecmap.h:
	* gst/ffmpegcolorspace/imgconvert.c:
	* gst/ffmpegcolorspace/imgconvert_template.h:
	* gst/playback/gstdecodebin.c:
	* gst/playback/gstplaybasebin.h:
	* gst/playback/gstplaybin.c:
	* gst/playback/gststreaminfo.h:
	* gst/tcp/gstfdset.c:
	* gst/tcp/gstfdset.h:
	* gst/tcp/gstmultifdsink.c:
	* gst/tcp/gstmultifdsink.h:
	* gst/tcp/gsttcp.h:
	* gst/tcp/gsttcpclientsrc.c:
	* gst/tcp/gsttcpclientsrc.h:
	* gst/tcp/gsttcpplugin.h:
	* gst/tcp/gsttcpserversink.c:
	* gst/tcp/gsttcpserversrc.c:
	* gst/typefind/gsttypefindfunctions.c:
	* gst/videorate/gstvideorate.c:
	* gst/videotestsrc/gstvideotestsrc.h:
	* gst/videotestsrc/videotestsrc.h:
	* sys/v4l/gstv4lcolorbalance.h:
	* sys/v4l/gstv4ltuner.h:
	* sys/v4l/gstv4lxoverlay.h:
	* sys/v4l/v4l_calls.h:
	* sys/v4l/videodev_mjpeg.h:
	* tests/check/elements/audioconvert.c:
	* tests/check/elements/audioresample.c:
	* tests/check/elements/audiotestsrc.c:
	* tests/check/elements/videotestsrc.c:
	* tests/check/elements/volume.c:
	* tests/examples/seek/scrubby.c:
	* tests/examples/seek/seek.c:
	  expand tabs
	  Original commit message from CVS:
	  expand tabs

2005-12-06 19:28:24 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* docs/libs/tmpl/gstaudio.sgml:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/libs/tmpl/gstgconf.sgml:
	* docs/libs/tmpl/gstmixer.sgml:
	* docs/libs/tmpl/gstringbuffer.sgml:
	* docs/libs/tmpl/gsttuner.sgml:
	* docs/libs/tmpl/gstxoverlay.sgml:
	  put back stability level
	  Original commit message from CVS:
	  put back stability level

2005-12-05 18:11:49 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* configure.ac:
	  back to HEAD
	  Original commit message from CVS:
	  back to HEAD

=== release 0.10.0 ===

2005-12-05 18:02:48 +0000  Thomas Vander Stichele <thomas@apestaart.org>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/libs/tmpl/gstcolorbalance.sgml:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-decodebin.xml:
	* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
	* docs/plugins/inspect/plugin-gnomevfs.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playbin.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-video4linux.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	  releasing 0.10.0
	  Original commit message from CVS:
	  releasing 0.10.0