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=== release 1.1.90 ===

2013-09-19  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  releasing 1.1.90

2013-09-19 09:45:18 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/cs.po:
	* po/nl.po:
	* po/pl.po:
	* po/uk.po:
	* po/vi.po:
	  po: Update translations

2013-09-11 14:27:02 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: dmabuf is not a singleton anymore
	  https://bugzilla.gnome.org/show_bug.cgi?id=707793

2013-09-16 13:53:45 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: do not do http requests in READY
	  HEAD requests to discover if the server is seekable shouldn't be done in
	  READY as it might lock the main thread that is doing the state change.
	  https://bugzilla.gnome.org/show_bug.cgi?id=705371

2013-09-18 16:32:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: reevaluate the current timer after timeout
	  When we trigger the timeout logic of a timer, reevaluate it because it is
	  possible that it still has the lowest timeout.

2013-09-18 16:31:26 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: don't update time when unscheduled
	  Don't try to estimate the current time when we got unscheduled.

2013-09-18 16:29:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: init packet spacing on first buffer
	  Already init the packet spacing variables on the first buffer so that we can
	  calculate the spacing on the second buffer already.

2013-09-18 15:08:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: fix comments

2013-09-18 14:57:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: push the lost event from the timer thread
	  Instead of pushing the lost event from the chain function, schedule a timeout
	  that will push the lost event from the timer thread. This avoid blocking the
	  upstream thread while we push and sync the event.

2013-09-18 14:23:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: add another test
	  The test is modified slightly because the late lost packets are only
	  generated now when a large gap is received.

2013-09-18 14:12:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: round gap duration to multiple of duration
	  Make sure the gap duration in the lost event is a multiple of the packet
	  duration.
	  Enable another test.

2013-09-18 12:29:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/Makefile.am:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: keep track of duration
	  Keep track of the estimated duration of missing packets and use it in the lost
	  event.
	  Enable another unit test

2013-09-18 11:59:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: handle large gaps with one lost event
	  When we have a large number of missing packets, generate one lost event for all
	  the packets that have no chance of being pushed out in time.
	  Fix and activate unit test for large gaps.

2013-09-18 11:56:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor lost event sending
	  Also make sure we only increment the expected seqnum and last
	  output timestamp.

2013-09-17 23:21:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor timeout triggers

2013-09-17 23:03:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: simplify the timeout code
	  Keep track of the current time in the timeout loop.
	  Loop over all timers and trigger all the expired ones, we can do this in the
	  same loop that selects the new best timer.

2013-09-17 23:01:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rearrange timer update code
	  Also update the timers when retransmission is disabled. We need to
	  do this because when we added LOST timers when we detected missing packets and
	  we need to remove those timers when the packet finally arrives.

2013-09-17 22:02:04 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/videomixer/Makefile.am:
	  videomixer: link to libm for maths stuff
	  Fixes undefined references to rint and pow on ubuntu
	  build bot.

2013-09-17 15:19:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: release lock on shutdown

2013-09-17 15:11:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/check/Makefile.am:
	  check: change for videomixer renamed orc file

2013-09-14 16:03:20 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: remove MAX_TOLERATED_LATENESS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707411

2013-09-16 15:54:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-H264-rtx.sh:
	  examples: we don't need the queue anymore

2013-09-16 15:53:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: use separate thread for timeouts
	  Use a separate thread for scheduling the timeouts instead of using the
	  downstream streaming thread that might block at any time.

2013-09-14 15:56:04 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: set first_ts to DTS for streams that have DTS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-14 15:55:22 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: make sure duration is a valid number for last buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-14 15:54:29 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: use segment.start or last buffer end time in case of missing DTS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-03 18:14:04 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/gstqtmux.c:
	  Revert qtmux: Use buffer PTS if DTS is not set"
	  This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707340

2013-09-16 11:03:06 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixerorc-dist.c:
	* gst/videomixer/videomixerorc-dist.h:
	  videomixer: Update orc generated files
	  https://bugzilla.gnome.org/show_bug.cgi?id=708131

2013-09-13 16:25:49 +0200  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Demux RTCP buffers from the RTP stream
	  If there are RTCP buffers in the RTP stream, process them as
	  RTCP. This way, we want receive streams following RFC 5761
	  https://bugzilla.gnome.org/show_bug.cgi?id=687657

2013-09-13 23:26:21 +1000  Jan Schmidt <thaytan@noraisin.net>

	* gst/rtp/gstrtpL24depay.c:
	  rtp: Remove bogus extra caps from L24 template.
	  The extra caps entry in the template was making it sometimes
	  get plugged for any dynamically allocated payload type.

2013-09-13 12:40:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: use PacketInfo for the sender
	  Avoid mapping the packet multiple times when sending RTP.

2013-09-13 12:22:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: store more in the PacketInfo
	  Store all info in the PacketInfo so that we can avoid mapping the packet
	  multiple times.

2013-09-13 11:32:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.h:
	  session: store more in the PacketInfo structure

2013-09-13 11:08:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	* gst/rtpmanager/rtpstats.h:
	  rtpbin: RTPArrivalStats -> RTPPacketInfo
	  Rename a structure because we are also going to use this for the sender
	  bits.

2013-09-13 10:55:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  source: small cleanups

2013-09-12 13:31:01 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: only update stop position if seek requests it
	  Check for GST_SEEK_TYPE_NONE for stop poistion and only update
	  the stop time if it is requested. Otherwise just maintain whatever
	  was stored at the segment
	  https://bugzilla.gnome.org/show_bug.cgi?id=707530

2013-09-13 08:53:25 +0200  Rico Tzschichholz <ricotz@ubuntu.com>

	* gst/rtp/Makefile.am:
	  rtp: Add missing headers tp fix make dist
	  In addition to a956a6ceb2deb87cc1361aee1d6626449f46dab2

2013-09-12 15:07:48 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Make sure we have enough data to read image tags
	  Thanks to iputinei for reporting this on IRC.

2013-09-12 15:01:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: handle segments with non-0 start
	  We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
	  transform it back to a buffer timestamp before pushing out the buffer.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931

2013-09-11 13:11:58 -0600  Seán de Búrca <leftmostcat@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix off-by-one in validation of UTF-8
	  https://bugzilla.gnome.org/show_bug.cgi?id=707933

2013-09-11 14:32:17 -0300  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Do not check if caps are empty when they are NULL
	  In the case the caps are actually NULL, we should just concider it the
	  same way as empty caps in that case.

2013-09-10 16:44:53 -0600  Seán de Búrca <leftmostcat@gmail.com>

	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	* gst/videomixer/videomixerorc-dist.c:
	* gst/videomixer/videomixerorc-dist.h:
	  videomixer: fix build if orc is not installed
	  https://bugzilla.gnome.org/show_bug.cgi?id=707886

2013-09-10 17:57:49 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Preserve seqnum when pushing seek upstream
	  After converting a seek from time to bytes, use the same seqnum
	  on the event that goes upstream

2013-09-05 00:17:16 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: track streams that are EOS on push mode to finish earlier
	  When the segment has a defined stop position, qtdemux should check
	  when streams reach this position and mark those as EOS. When all
	  streams are EOS it will return GST_FLOW_EOS to upstream to allow
	  the pipeline to finish instead of continuously consume buffers
	  from upstream that are not useful for the segment.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707530

2013-09-04 15:34:35 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: preserve stop of segment when doing seeks in push mode
	  When handling seeks in push mode, qtdemux converts the seek to bytes
	  and pushes upstream. It needs to keep track of the seek and the
	  subsequent segment to be able to map them back to the requested
	  seek time and properly preserve the segment stop of the seek.
	  This is done by using the start offset in bytes of the seek,
	  that should be the same of the segment from upstream. And this
	  is also backwards compatible with what qtdemux already was using.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707530

2013-07-26 19:40:53 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2pad.h:
	  videomixer: Add colorspace conversion
	  https://bugzilla.gnome.org/show_bug.cgi?id=704950

2013-08-06 15:38:39 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't send reconfigure event when formats or PAR are different
	  It is racy with multiple pads.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704950

2013-07-25 13:49:57 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/Makefile.am:
	* gst/videomixer/blend.c:
	* gst/videomixer/blendorc.orc:
	* gst/videomixer/gstcms.c:
	* gst/videomixer/gstcms.h:
	* gst/videomixer/videoconvert.c:
	* gst/videomixer/videoconvert.h:
	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixerorc.orc:
	  videomixer: Bundle private copies of videoconvert code
	  Ideally, this would be part of libgstvideo.
	  Prefixes videoconvert symbols with videomixer_.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704950

2013-08-22 00:03:48 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: Use newly #defined metadata names.

2013-09-09 15:11:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: only wait if we flushed
	  Only wait for the STREAM_LOCK when we flushed something when sending
	  a command for PAUSED or PLAYING.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611

2013-09-09 15:09:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: return when a flush was issued
	  Make gst_rtspsrc_loop_send_cmd() return TRUE when the current
	  action has been flushed

2013-09-09 11:16:40 +0200  David Holroyd <dave@badgers-in-foil.co.uk>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpL24depay.c:
	* gst/rtp/gstrtpL24depay.h:
	* gst/rtp/gstrtpL24pay.c:
	* gst/rtp/gstrtpL24pay.h:
	* tests/check/elements/rtp-payloading.c:
	  rtp: add L24 pay and depayloader
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734

2013-09-09 14:46:42 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix missing condition in previous commit

2013-09-09 14:44:58 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Also fix strides for other semi-planar video formats

2013-09-09 14:41:42 +0200  Andreea Fulger <andreea.fulger@parrot.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: Fix stride for NV12/NV21
	  https://bugzilla.gnome.org/show_bug.cgi?id=707758

2013-09-07 16:37:03 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: fix leaking buffer and caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=707688

2013-09-05 19:46:37 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/udp/gstudpsrc.c:
	  udpsrc: fix build on win32
	  gstudpsrc.c:855:15: error: #if with no expression

2013-09-04 15:50:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: handle unseekable streams
	  Handle streams that we can't seek in and ignore them in the
	  seek logic.

2013-09-04 15:25:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/avi/gstavidemux.c:
	  avidemux: only check video compression for video streams
	  Or else we might deref a stream with a NULL strf.vids and segfault

2013-06-18 13:27:20 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/atoms.c:
	* gst/isomp4/fourcc.h:
	* gst/isomp4/ftypcc.h:
	* gst/isomp4/gstrtpxqtdepay.c:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add support for the avc3 sample entry format of the AVC file format
	  Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
	  structure for fragmented MP4 called "avc3". The principal difference
	  between AVC1 and AVC3 is the location of the codec initialisation
	  data (e.g. SPS, PPS). In AVC1 this data is placed in the initial
	  MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data
	  goes in the first sample of every fragment (i.e. the first sample in
	  each mdat box).  The principal reason for avc3 is to make it easier
	  for client implementations, because it removes the requirement to
	  insert the SPS+PPS in to the decoder pipeline every time there is a
	  representation change.
	  This commit adds support for the "avc3" atom, which is almost identical
	  to the "avc1" atom, except it does not contain any SPS or PPS data.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702004

2013-09-04 00:27:50 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: Don't set EOS to FALSE when the collectpad *is* EOS
	  https://bugzilla.gnome.org/show_bug.cgi?id=707238

2013-09-03 17:32:41 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: cleanup on error after state change
	  https://bugzilla.gnome.org/show_bug.cgi?id=707229

2013-09-03 11:23:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/udp/gstudpsrc.c:
	* gst/udp/gstudpsrc.h:
	  udpsrc: Bind to multicast addresses on non-Windows systems
	  On Windows it's not possible to bind to a multicast address
	  but the OS will make sure to filter out all packets that
	  arrive not for the multicast address the socket joined.
	  On Linux and others it is necessary to bind to a multicast
	  address to let the OS filter out all packets that are received
	  on the same port but for different addresses than the multicast
	  address
	  And deprecate the multicast-group property and replace it with the
	  address property.
	  https://bugzilla.gnome.org/show_bug.cgi?id=707042

2013-09-03 10:10:01 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Free GstBaseParseFrame if pushing a header failed

2013-09-02 16:02:37 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Refactor address resolval into its own function

2013-09-02 23:00:29 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/replaygain/gstrganalysis.c:
	  replaygain: fix taglist leak in rganalysis
	  And add some FIXMEs.

2013-09-02 22:50:58 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/rganalysis.c:
	  tests: rganalysis: rename function for clarity

2013-03-18 14:32:07 +0100  Christoph Reiter <reiter.christoph@gmail.com>

	* tests/check/elements/rganalysis.c:
	  tests: fix skipped rganalysis tests
	  In 0.10 elements would post tag messages on the bus
	  directly, and rganalysis would only post a tag message
	  when it changed tags. In 1.0, only sinks post tag
	  messages when they receive the serialised tag event.
	  This means that we get an additional tag message on
	  the bus now where we didn't expect one before.
	  https://bugzilla.gnome.org/show_bug.cgi?id=695090

2013-09-02 11:46:52 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Properly propagate downstream flow returns upstream
	  https://bugzilla.gnome.org/show_bug.cgi?id=707229

2013-09-01 21:18:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/shout2/gstshout2.c:
	* gst/avi/gstavi.c:
	* gst/isomp4/isomp4-plugin.c:
	* gst/rtsp/gstrtsp.c:
	* sys/sunaudio/gstsunaudio.c:
	* sys/v4l2/gstv4l2.c:
	  Don't use setlocale in plugins()
	  Only apps should call setlocale(), not libraries.

2013-08-29 13:15:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpmpvpay.c:
	  rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay
	  RTP buffer allocation should not be done with padding for the specific MPEG2
	  header as the padding is done at the end of the buffer and the last byte is
	  the size of the padding.
	  https://bugzilla.gnome.org/show_bug.cgi?id=706970

2013-08-28 10:51:32 +0200  Bernhard Miller <bernhard.miller@streamunlimited.com>

	* gst/autodetect/gstautovideosink.c:
	* gst/autodetect/gstautovideosink.h:
	  autovideosink: add sync property
	  https://bugzilla.gnome.org/show_bug.cgi?id=706955

2013-08-28 07:15:00 +0200  Bernhard Miller <bernhard.miller@streamunlimited.com>

	* gst/autodetect/gstautoaudiosink.c:
	* gst/autodetect/gstautoaudiosink.h:
	  autoaudiosink: introduce sync property
	  https://bugzilla.gnome.org/show_bug.cgi?id=706955

2013-08-27 17:33:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: push buffers after segment stop until reaching a keyframe
	  This should make decoders able to precisely push buffers until the stop
	  time in case they need the next keyframe to do it.
	  Also, according to gst_segment_clip, it should only push a buffer that
	  the starting ts is strictly smaller than the segment stop, so we change
	  the min < comparison for <=

2013-08-28 13:26:47 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

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=== release 1.1.4 ===

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2013-08-28 12:52:25 +0200  Sebastian Dröge <slomo@circular-chaos.org>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/audiofx/audiopanoramaorc-dist.c:
	* win32/common/config.h:
	  Release 1.1.4

2013-08-28 12:52:16 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
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2013-08-28 12:32:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/pt_BR.po:
	  po: update translations

2013-08-27 15:25:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: remove framerate restriction
	  Remove the framerate restriction on the caps.

2013-08-27 09:38:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: only update next check time when reconsidering
	  Don't update the next RTCP check time in all cases but only when we
	  reconsidered. This avoids delaying sending a full RTCP packet when we
	  are doing early feedback.

2013-08-27 09:37:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add more debug

2013-08-27 09:34:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  jitterbuffer: fix types of the retransmission event

2013-08-27 09:33:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: only timeout EXPECTED timers on gap
	  Only timeout the EXPECTED timers when we detect a large seqnum gap.

2013-08-26 13:47:53 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  configure.ac: Don't set BZ2_LIBS if bz2 is not found

2013-08-26 11:50:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtsession: fix locking
	  We need to take the session lock when getting and manipulating the
	  source.

2013-08-26 11:50:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: add some more debug

2013-08-20 22:12:03 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: don't send flush_stop twice.
	  If we get flush start and a seek we need to only send flush_stop once.
	  More info at #706441

2013-08-23 15:56:43 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: propagate discont

2013-08-23 15:49:47 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: remove dynamic sourcpads when going from PAUSED to READY

2013-08-23 15:29:28 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last
	  https://bugzilla.gnome.org/show_bug.cgi?id=637754

2013-08-23 15:47:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtxqueue: add property to configure queue size

2013-08-23 12:07:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-H264-rtx.sh:
	* tests/examples/rtp/server-VTS-H264-rtx.sh:
	  tests: add retransmission example

2013-08-23 11:55:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: proxy jitterbuffer do-retransmission property

2013-08-23 11:17:45 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/avi/gstavimux.c:
	  avimux: unmap the correct buffer
	  The audio buffer was mapped so unmap it and not the video buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=706642

2013-08-18 23:32:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Add property to find out the device currently in use
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:31:15 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: De-duplicate code to get the current sink input info
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 22:27:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Implement changing the device while playing
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:32:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: Add property to find out the device currently in use
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:31:15 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: De-duplicate code to get the current source output info
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 22:27:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Implement changing the device while playing
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-22 14:55:14 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  configure: Fix bz2 configure check for Windows
	  Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2.
	  https://bugzilla.gnome.org/show_bug.cgi?id=465924

2013-02-22 20:57:00 +0900  Akihiro Tsukada <atsukada@users.sourceforge.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesink: Add support for AAC pass-through
	  https://bugzilla.gnome.org/show_bug.cgi?id=694445

2013-06-24 17:29:37 +0200  Kishore Arepalli <kishore.arepalli@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: crashes if any property changes during playback when location property is not set
	  https://bugzilla.gnome.org/show_bug.cgi?id=702988

2013-08-21 14:54:26 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.h:
	  pulse: Share static caps definition between src and sink
	  The src was also missing 24-bit sample formats

2013-08-21 16:53:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtx: various improvements
	  Use locking
	  Don't push from the event handler, collected packets in a queue and push from
	  the chain function.
	  Clear queues on shutdown.

2013-08-21 16:50:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  session: generate events correctly
	  Do correct shifting of the bitmask for lost packets.

2013-08-21 16:47:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpmanager.c:
	  rtp: register rtx element better

2013-08-21 16:32:50 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others
	  Probably fixes
	  https://bugzilla.gnome.org/show_bug.cgi?id=705477

2013-08-21 13:03:34 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: don't ignore return value from _finish_frame()
	  gst_video_encoder_finish_frame() will return FLOW_OK here if
	  there's no output buffer.

2013-08-21 12:56:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	  jpegdepay: add some more debug

2013-08-21 12:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	  rtpgstdepay: only push events when they changed
	  Keep track of the STREAM_START and TAG events and only push them
	  when they changed.

2013-08-21 10:52:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: taglists should not be merged in 1.0

2013-08-21 10:28:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: flush on FLUSH_STOP event

2013-08-21 10:03:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: reset on state change
	  Do full reset on state change to READY

2013-08-21 09:55:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: reset on FLUSH_STOP
	  Clear the adapter and pending buffer list on FLUSH_STOP.

2013-08-21 09:39:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: don't use clock for config interval
	  We can't use the clock to time our config-interval because we are not
	  live (or there might not be a clock or the clock might not be running).
	  Instead just simply take the timestamp diff.

2013-08-21 09:33:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.h:
	  rtpgstay: don't use // comments

2013-08-08 11:55:22 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix response argument in handle-request signal

2013-08-08 11:54:41 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add sdes property and proxy it to rtpbin

2013-08-07 09:47:35 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  Send a stream-start whenever we send tags This is to make sure tags are cleared on the client if the stream-start was previously lost, otherwise, the client may end up with a merged taglist of multiple songs

2013-07-25 21:12:05 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval This is useful in case the packet containing the inlined caps was lost or if new client joins an already running RTP stream and they missed the previous tag events. This also makes the payloader keep a list of merged tags so the retransmitted tag event contains all previously received. A STREAM_START event will flush the list of tags.

2013-07-25 21:10:10 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time

2013-07-25 21:03:34 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps

2013-07-25 20:54:50 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList This is necessary to fix event/caps sending. If we send a STREAM_START packet, it will cause an error because the stream didn't receive its caps and new-segment events, so we must wait for the first buffer before sending the stream-start event buffer. However, the caps will be sent at the same time and so the 'inline caps' will be set for the event. We need to be able to payload individual packets (data, caps or events) and only send them when we call flush.

2013-07-25 17:56:38 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START

2013-07-25 17:52:16 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3

2013-08-20 14:36:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: handle EOS
	  When the queue is empty, and we received EOS, pause and push an EOS
	  event downstream.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387

2013-08-20 10:26:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update docs

2013-08-20 10:25:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update all timers
	  Keep looping over all registered timers so that we can mark them lost instead of
	  stopping as soon as we find the timer for the current seqnum.

2013-08-20 08:55:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: remove unused variables

2013-08-19 21:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reorganize timer handling
	  Restructure handling of incomming packet and the gap with the expected seqnum
	  and register all timers from the _chain function.
	  Convert a timer to a LOST packet timer when the max amount of retransmission
	  requests has been reached.

2013-08-19 21:37:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor packet spacing calculation

2013-08-19 21:34:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: keep track of last seqnum and dts

2013-08-19 21:29:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanups

2013-08-19 21:21:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reset retransmission timers in add/reschedule
	  Reset the retransmission timers when adding and rescheduling a timer.

2013-08-19 21:12:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rename variables for packet spacing

2013-08-19 14:58:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: remove lost timer when we get the packet
	  When we receive a packet, also remove the LOST timer for it.

2013-08-19 14:56:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: expected seqnum must increase
	  Only update the expected seqnum when it is bigger than the previous expected
	  seqnum.

2013-08-19 14:55:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add more debug

2013-08-12 16:15:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtxqueue: add retransmission queue element

2013-08-12 14:53:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add some docs

2013-08-06 16:29:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: handle NACK feedback and generate events
	  Handle and parse the feedback NACK packets and generate a Retransmission
	  event for each NACKed packet

2013-08-19 13:19:42 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add forward declaration for gst_v4l2_object_get_format_list

2012-10-22 17:58:07 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: De-duplicate caps probing between src and sink

2013-08-13 17:32:17 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/Makefile.am:
	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulseprobe.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulse: Remove unused GstPulseProbe

2013-08-19 12:46:45 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tunerchannel.c:
	* sys/v4l2/tunernorm.c:
	  v4l2: Use G_DEFINE_ macros for added thread safety

2013-08-17 11:28:13 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: Do not send flush_stop ourself after a flush_start
	  When we receive a flush_start, we should wait for the next flush_stop
	  and foward it, not create a flush_stop ourself.

2013-08-16 17:10:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: init debug category early
	  Init the debug variable when we register the element because it is also used by
	  the payloader element when it calls the add_sps_pps method.

2013-08-16 13:26:28 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/flac/gstflacenc.c:
	  flacenc: Properly set headers via the base class instead of just pushing them downstream
	  Prevents buffers from being send before the caps and segment events.

2013-08-15 10:59:10 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check denominator isn't zero before scaling duration.
	  When gst_qtdemux_configure_stream sets fps_d, check that n_samples is
	  non-zero before using it as a denominator to scale the stream duration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=706076

2013-08-15 15:08:05 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  ext: Use new flush vfunc of video codec base classes and remove reset implementations

2013-08-14 16:19:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: forward flush before stopping dataflow
	  First forward the flush event and then stop our loop function.

2013-08-14 13:10:32 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	  configure: require libsoup >= 2.38
	  Bump libsoup requirement for newer API used, like headers_get_one().
	  2.38 is from early 2012 and is in linen with our GLib requirement.

2013-08-14 11:54:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/soup/gstsouphttpsrc.c:
	  soup: don't use deprecated soup_message_headers_get() API

2013-08-13 17:44:50 +0200  Edward Hervey <edward@collabora.com>

	* .gitignore:
	  .gitignore: Ignore files from automake test-driver

2013-08-12 15:28:34 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Use the SPS/PPS handling function from the depayloader
	  Remove duplicated copies
	  https://bugzilla.gnome.org/show_bug.cgi?id=705553

2013-08-12 15:26:08 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: Make the SPS/PPS deduplication function generic
	  Make it not touch any internals of the depayloader
	  https://bugzilla.gnome.org/show_bug.cgi?id=705553

2013-08-13 14:09:20 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: allow conversion from raw AAC to ADTS
	  This patch will prepend ADTS headers to raw AAC audio frames, allowing
	  upstream elements to link to decoders that only support AAC in ADTS format.
	  Note that no error correction bits are added to ADTS frames in this code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=615740

2013-08-13 12:44:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Only free GCheckSum after its last usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=705760

2013-08-13 12:02:29 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix critical setting a NULL uri redirection

2013-07-13 01:50:56 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: add redirection to the URI query

2013-07-31 10:42:07 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: elst should offset samples instead of buffers
	  The current approach where buffers are offset is not ideal, as during seek
	  and loop current time is compared to sample times.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700264

2013-08-07 19:32:07 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	* tests/check/elements/videomixer.c:
	  videomixer: Send EOS if buf_end >= segment.stop
	  That means the whole segment is already played, and we are sure we
	  are EOS at that point.
	  Also handle segment seeks, and do not send EOS in that case.

2013-08-04 14:40:38 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: send proper stream_start event
	  https://bugzilla.gnome.org//show_bug.cgi?id=705449

2013-08-08 11:51:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't print warnings during flushing and stop as soon as possible
	  https://bugzilla.gnome.org//show_bug.cgi?id=705442

2013-08-07 11:14:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: mark key frames and delta frames properly
	  https://bugzilla.gnome.org/show_bug.cgi?id=705550

2013-08-05 23:23:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add NACK feedback in RTCP

2013-08-05 23:22:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  source: add methods to register NACK
	  Add a method to register a missing packet for an ssrc along with
	  methods to get the missing packets and clear them.

2013-08-04 23:05:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: handle Retransmission event and schedule NACK
	  Handle the retransmission event from downstream and use it to schedule a NACK
	  request.

2013-08-05 23:20:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: pass data to remove func
	  Pass the data to the remove function because we are going to deref it when there
	  is pli or fir.

2013-08-06 15:28:50 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix compilation

2013-08-06 15:17:44 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE

2013-08-06 11:58:38 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Make sure to send EOS if the buffer end time equals the segment end time
	  Otherwize EOS never gets sent in that particular case.

2013-08-05 08:49:50 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: Ensure src caps are writable
	  In some cases the src caps determined by goom weren't writable, causing
	  a bunch of assertion failures and failed caps. Fixed by always
	  explicitely making the caps writable
	  https://bugzilla.gnome.org/show_bug.cgi?id=705475

2013-08-04 23:18:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: use common send_rtcp method
	  Reuse the send_rtcp method that already asks for the current time when
	  requesting a keyframe.

2013-08-04 23:12:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: Don't use ClockTimeDiff for unsigned delays

2013-08-04 16:52:15 +0200  Edward Hervey <edward@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use buffer PTS if DTS is not set
	  Avoids ending up with completely bogus scaled duration/pts when new
	  buffers have invalid DTS.

2013-08-04 14:32:47 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/souphttpsrc.c:
	  tests: skip https test if there's no TLS support in soup/glib

2013-08-04 11:20:41 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtsp/gstrtpdec.c:
	  rtpdec: use generic marshaller

2013-08-04 10:52:33 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* sys/v4l2/.gitignore:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2-marshal.list:
	* sys/v4l2/tuner-marshal.list:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tuner.h:
	* win32/MANIFEST:
	* win32/common/tuner-enumtypes.c:
	* win32/common/tuner-enumtypes.h:
	* win32/common/tuner-marshal.c:
	* win32/common/tuner-marshal.h:
	  v4l2: remove unused enumtypes and use generic marshaller

2013-08-04 10:47:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* gst/udp/.gitignore:
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-enumtypes.h:
	* win32/common/gstudp-marshal.c:
	* win32/common/gstudp-marshal.h:
	  udp: remove unused marshal and enumtypes files

2013-08-04 09:38:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* gst/rtpmanager/.gitignore:
	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpsession.c:
	* win32/MANIFEST:
	* win32/common/gstrtpbin-marshal.c:
	* win32/common/gstrtpbin-marshal.h:
	  rtpmanager: use generic marshaller

2013-08-04 00:13:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: send event in right direction

2013-08-02 17:38:34 -0700  David Schleef <ds@schleef.org>

	* configure.ac:
	* tests/check/Makefile.am:
	  tests: create/remove orc directory at proper time
	  Before automake creates .deps directories, and during distclean.

2013-08-03 00:25:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add FIR and PLI like other RTCP packets
	  Add the FIR and PLI packets like the other RTCP packet instead of from the
	  on-sending-rtcp default signal handler.

2013-08-02 17:22:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix property ranges

2013-08-02 16:42:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: push retransmission events

2013-08-02 14:12:16 +0200  Lubosz Sarnecki <lubosz@gmail.com>

	* configure.ac:
	  build: add subdir-objects to AM_INIT_AUTOMAKE
	  Fixes warnings with automake 1.14
	  https://bugzilla.gnome.org/show_bug.cgi?id=705350

2013-08-02 14:54:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add support for retransmission retry
	  When we didn't receive a packet after requesting retransmission, retry
	  asking for retransmission for a certain period.

2013-08-02 14:19:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add properties
	  Add properties to control retransmission parameters

2013-08-02 12:44:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: use corrected timeout when rescheduling
	  When we recalculate the timeout, use the corrected timeout value depending on
	  the timer type.

2013-08-02 12:43:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update timers after queueing
	  Else we might update the timer needlessly for duplicates.

2013-08-02 12:42:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: move method up

2013-08-02 06:28:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanup

2013-08-01 23:26:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: unschedule old expected packets
	  When we receive a new packet, unschedule old outstanding packets when their
	  seqnum is too far away.

2013-08-01 23:29:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor timer update

2013-08-01 23:24:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update timers when removing
	  Update the timers when we remove a timer.
	  Handle canceled timers, make them unschedule the current timer and
	  trigger the timeout code.

2013-08-01 23:22:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix typo

2013-08-01 15:40:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve timeout management
	  If we change the seqnum of an existing timer and we were waiting for
	  that timer, unschedule it. If we change the timeout of an existing timer and we
	  were waiting on it, only unschedule when the new time is smaller.

2013-08-01 15:05:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: install timer for expected arrival
	  Install a timer that is triggered when the expected arrival time of a packet
	  expired.

2013-08-01 14:56:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve unschedule of timers
	  Conflicts:
	  gst/rtpmanager/gstrtpjitterbuffer.c

2013-08-01 12:21:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: move code around

2013-08-01 12:07:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: estimate inter packet spacing
	  When we see two packets with consecutive seqnums and a different RTP time, use
	  the DTS difference as the inter packet spacing estimate.

2013-08-01 12:01:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: keep track of current timeout

2013-08-01 11:49:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: cleanup timer handling

2013-08-01 11:40:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reset is only possible with a GAP

2013-08-01 11:29:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: operate on DTS
	  Make the jitterbuffer schedule the timeouts based on the DTS instead
	  of the PTS. This makes it all smoother with reordered frames and gives
	  the decoder time to reorder the frames in time.

2013-08-01 11:14:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rename timout variable

2013-07-31 17:08:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanup

2013-07-31 16:59:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: block output in paused or buffering

2013-07-31 16:59:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: store pts in timer
	  Only store the pts in the timer so that we can both do timeouts with timings on
	  the input and output of the jitterbuffer.

2013-07-30 23:14:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor jitterbuffer
	  Refactor the jitterbuffer code. Make separate function for peeking a buffer,
	  pushing the next buffer, waiting for timeouts and handling the timeouts.
	  The main loop now tries to push as many buffers as it can until it runs out of
	  buffers or when it detects a seqnum discont. Then it will wait for some event to
	  happen before attempting to push more buffers.
	  Make methods to register timeouts in an array. These timeouts are registered
	  when we detect a missing packet, sync for the first packet or when we find an
	  estimation for the end-of-stream.
	  This greatly simplifies and clarifies the code and also makes it possible to
	  register more complicated timeout schemes later.

2013-07-30 18:52:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: use NULL to ignore percent
	  If we pass NULL to pop and push we ignore the percent result.

2013-07-30 07:00:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor
	  Move eos estimation into separate function

2013-07-30 14:28:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: don't leak stream_id string
	  https://bugzilla.gnome.org/show_bug.cgi?id=705142

2013-07-29 19:53:52 +0100  Tim-Philipp Müller <tim@centricular.net>

	* po/LINGUAS:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/ja.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sl.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: update translations

2013-07-29 19:48:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/.gitignore:
	  tests: ignore new test binaries

2013-07-29 14:47:49 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

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=== release 1.1.3 ===

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2013-07-29 13:42:18 +0200  Sebastian Dröge <slomo@circular-chaos.org>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.1.3

2013-07-29 13:42:05 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
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2013-07-29 12:12:41 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	* gst/flv/gstflvdemux.c:
	* gst/isomp4/qtdemux.c:
	* gst/matroska/matroska-demux.c:
	  gst: Don't swap start/stop for negative rates in the SEGMENT query

2013-07-29 11:18:40 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check for data size when parsing h264 codec data from strf atom

2013-07-29 10:53:54 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Implement SEGMENT query

2013-07-29 10:53:47 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Implement SEGMENT query

2013-07-29 10:50:59 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	  avidemux: Implement SEGMENT query

2013-07-27 18:10:22 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: Support H264 fourcc
	  https://bugzilla.gnome.org/show_bug.cgi?id=704996

2013-07-28 18:09:33 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/flac/gstflacenc.c:
	  flacenc: Fix handling of image tags
	  The caps should be used to get the mimetype and there is
	  only an info structure for the GstSample if the image-type
	  is not NONE.

2013-07-28 18:04:32 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/flac/gstflacenc.c:
	  flacenc: Don't crash if there is no image tag information
	  https://bugzilla.gnome.org/show_bug.cgi?id=705018

2013-07-28 17:38:56 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix duration reporting in push mode
	  https://bugzilla.gnome.org/show_bug.cgi?id=700933

2013-07-28 17:32:27 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	  avidemux: Don't forget unmapping and unreffing buffer

2013-07-26 21:06:17 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: unmap buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=704951

2013-07-26 22:31:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: don't make buffer writable prematurely
	  There is no reason to make the SR buffer writable at this point. This is better
	  delayed until needed.

2013-07-26 22:25:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: ignore RTCP for inactive sources

2013-07-26 22:25:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: small cleanup

2013-07-26 17:17:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.h:
	  session: handle partial RTCP report blocks
	  When we have more SSRCs to report than what fit in an RTCP packet, use a
	  generation counter to make sure all of them end up in a packet eventually.

2013-07-26 17:23:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: create SSRC before doing session cleanup
	  Make the internal source before we do session cleanup

2013-07-26 17:21:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: reorganize the report block code

2013-07-26 16:02:01 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: fix memory leak in check_subtitle_buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=704921

2013-07-26 14:21:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: refactor active and sender checks

2013-07-26 12:06:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: remove internal sources on timeout
	  When an internal source times out and becomes a receiver, remove it.

2013-07-26 11:47:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: create an internal source for RTCP
	  When we need to do RTCP and we don't have an internal source yet,
	  make one.

2013-07-26 10:47:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	  session: remove old code to change SSRC
	  Remove code used to change the SSRC after a collision. We now send
	  a RECONFIGURE event upstream to make the upstream element change the SSRC.

2013-07-26 10:42:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  source: don't update packet SSRC
	  Remove the code to update the SSRC in packets, it can never be called now that
	  we always use a source with matching packet SSRC.

2013-07-26 10:24:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: delay allocation of internal source
	  Allocate the internal source when we receive a caps with the SSRC or when we see
	  a buffer with the SSRC.

2013-07-26 10:00:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  session: generate reconfigure on collision
	  When we detect a collision, change the SSRC that we suggest upstream
	  and trigger RECONFIGURE. This should make upstream select a new SSRC.

2013-07-26 09:37:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: produce RTCP for all internal sources
	  Loop over all the internal sources and produce RTCP. We also need
	  to queue the RTCP packets and send them when we are finished.

2013-07-26 01:40:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: deprecate internal source and ssrc properties
	  Deprecate the internal source and internal ssrc properties. There might
	  be more than one internal source.

2013-07-26 01:29:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: internal sources don't use probation

2013-07-26 01:24:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	  session: give caps to session
	  Let the session parse the caps and update its SSRC when needed.

2013-07-26 01:14:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: make method to suggest available SSRC
	  Make a method to suggest the best available SSRC. This is the SSRC of the last
	  created internal source and is used to instruct upstream to produce this
	  SSRC.

2013-07-26 01:01:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: keep SDES and set on new internal sources
	  Keep track of the SDES ourselves and set it on all newly created
	  internal sources.

2013-07-26 00:48:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: make method to make internal sources
	  Add a method to obtain an internal source and use it to create
	  our internal source

2013-07-26 00:29:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpstats.h:
	  session: count internal sources and how many are senders

2013-07-26 00:14:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: separate BYE marking and scheduling
	  First mark sources with BYE and then schedule the BYE RTCP message.

2013-07-25 23:56:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: get SSRC from RTCP packet itself
	  Get the SSRC from the RTCP packet instead.

2013-07-25 23:51:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: fix bandwidth calculation
	  We iterate over all sources and the internal one is also in the
	  hashtable so avoid adding it twice.

2013-07-25 23:38:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add some docs

2013-07-25 23:11:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: Rearrange RTCP reporting a little
	  Make a function to generate an RTCP packet for a source, pass the source as a
	  parameter.
	  Move timeout of collisions to session cleanup phase.

2013-07-25 22:39:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: move check for is_early around
	  Move the check for the early RTCP to where it is needed and used.

2013-07-25 17:35:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: parse packet outside of the session lock

2013-07-25 17:34:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: do nicer checks for internal sources

2013-07-25 17:15:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  session: let source keep track if it sent BYE

2013-07-25 17:06:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	  source: reset more

2013-07-25 16:49:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  source: also use the source for bye_reason
	  Store the BYE reason in our internal source object. Rename the methods on the
	  source object a little because now the BYE can be received in RTCP or
	  set when the session wants to send BYE.

2013-07-25 16:24:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  session: configure sdes with structure only
	  Remove code to configure the SDES with methods and types, only
	  allow configuration with GstStructure

2013-07-25 15:56:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: refactor add and find source
	  Make functions to find and add a source to the hashtable.

2013-07-25 15:43:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: remove source from sync_rtcp
	  We don't need to know the sender source of the session in the
	  callback, the SR packet is for all participants in the session.

2013-07-24 14:18:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add some more debug

2013-07-15 17:11:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/Makefile.am:
	* gst/audioparsers/gstaacparse.c:
	* gst/audioparsers/gstaacparse.h:
	  aacparse: allow conversion from ADTS to raw AAC
	  Some muxers (eg, qtmux) only support raw AAC, so this allows linking
	  an encoder that outputs ADTS only to those muxers.
	  The conversion is simple (omit the first 7 or 9 bytes of the frame),
	  but has to be done in pre_push instead of handle_frame as 1.0 does
	  not seem to allow skipping bytes there as 0.10 used to.
	  Other conversions are not supported (yet).

2013-07-15 17:15:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix object_type parsing off-by-one in ADTS frame
	  According to http://wiki.multimedia.cx/index.php?title=ADTS,
	  the value stored in ADTS headers is one less than the object
	  type of the AAC stream.
	  A look at ffmpeg shows it also adds 1 to the value read off
	  the ADTS header.
	  Note that this might break other things that happen to have
	  an inverse off by one to match the existing code.

2013-07-25 11:13:01 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/avi/gstavidemux.c:
	  avidemux: fix seqnum handling for seeks
	  Use the same seqnum as the seek for flushes/segments that are
	  caused by the seek. Also do the same for segment events
	  Fixes #676242

2013-07-25 01:39:58 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: fix seqnum handling for seeks
	  Use the same seqnum as the seek for flushes/segments that are
	  caused by the seek. Also do the same for segment events
	  Fixes #676242

2013-07-25 01:11:31 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: correctly handle seqnum for seeks and segments
	  Use the same seqnum on messages and events for derived events.
	  Fixed for flushes / stream-start / segment after a seek, and segment
	  after a segment.
	  Fixes #676242

2013-07-12 20:01:42 +0200  Arnaud Vrac <avrac@freebox.fr>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: always ignore HEAD errors
	  https://bugzilla.gnome.org/show_bug.cgi?id=704241

2013-07-25 14:26:07 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: Clean up reset/start/stop handling

2013-07-25 14:13:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegdec.h:
	  jpegdec: Use base class error handling function instead of replicating it here

2013-07-25 14:12:56 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Clean up handling of reset/start/stop

2013-07-25 10:41:22 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* tests/files/id3-407349-1.tag:
	* tests/files/id3-407349-2.tag:
	* tests/files/id3-447000-wcop.tag:
	  tests: fix test ID3 tags up not to rely on dodgy typefinding code
	  Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix
	  the typefinder.
	  https://bugzilla.gnome.org/show_bug.cgi?id=681368

2013-07-25 08:22:45 +0200  Alessandro Decina <alessandro.d@gmail.com>

	* sys/osxaudio/gstosxaudiosink.c:
	  osxaudiosink: intersect the probed caps with the filter passed to get_caps()

2013-07-24 14:17:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	  bin: fix compilation

2013-07-24 12:42:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawdepay.c:
	  vrawdepay: fix UYVP format

2013-07-24 12:41:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  vrawpay: fix UYVP format

2013-07-24 12:41:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvrawpay.c:
	  vrawpay: fix caps

2013-07-24 10:49:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: fix locking
	  Take the lock earlier so that we do things that follow with the right
	  locking.

2013-07-23 17:40:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: don't use invalid times in RTCP timeouts
	  An invalid timeout can be calculated when we disabled RTCP by setting the
	  bandwidth to 0. Make sure all code can handle this case.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626

2013-07-23 17:38:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: lock session when changing bandwidth
	  Take the session lock when changing the bandwidth properties so that we don't
	  end up with inconsistent behaviour.

2013-07-23 17:37:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: reset some RTCP variables
	  The early_send time was set to 0 and always triggering an early RTCP packet.

2013-07-23 15:03:31 +0200  Edward Hervey <edward@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add all the mpeg XDCAM variants
	  This should cover all known XDCAM variants (which are all mpeg2 video)
	  Fixes #672227

2013-07-03 18:41:42 +0200  Carlos Rafael Giani <dv@pseudoterminal.org>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: added custom downstream sync event
	  rtpbin can now send a custom in-band downstream event which informs
	  downstream that the bin has received an RTCP SR packet. This is useful
	  for applications which want to drop the initial unsynchronized received
	  RTP packets.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560
	  Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>

2013-07-22 18:00:16 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: fix on-the-fly changing of "mode" and "fields" properties
	  We call setcaps() to reconfigure ourselves, but we need to pass
	  the current *sink* caps, not the source caps then. Also fix a
	  caps leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=641599

2013-07-22 15:23:39 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Add support for group-id in the stream-start event

2013-07-22 15:23:20 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Add support for group-id in the stream-start event

2013-07-22 15:23:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtpmanager/gstrtpsession.c:
	  rtpsession: Add support for group-id in the stream-start event

2013-07-22 15:22:55 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	  matroskademux: Add support for group-id in the stream-start event

2013-07-22 15:22:47 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Add support for group-id in the stream-start event

2013-07-22 15:22:36 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Add support for group-id in the stream-start event

2013-07-22 15:22:16 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/avi/gstavidemux.c:
	* gst/avi/gstavidemux.h:
	  avidemux: Add support for group-id in the stream-start event

2013-07-22 15:21:49 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/dv/gstdvdemux.c:
	* ext/dv/gstdvdemux.h:
	  dvdemux: Add support for group-id in the stream-start event

2013-07-19 22:59:15 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: use gst_util_uint64_scale*_round.
	  There could be a case where:
	  1) you do a new set_caps after buffers have been processed.
	  2) ts_offset gets set to a different value, eg 0.033333333
	  3) your pads get EOS, but the check dor that doesn't work
	  because you use ts_offset + a truncated value < segment.stop
	  4) so in the next collected, you end up comparing for example:
	  0.9999999999 > 1., which is false and means you don't send EOS.
	  Also adds scale_round in two other places where it potentially could
	  have caused problems.

2013-07-15 17:55:19 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: Add WRLE support

2013-07-19 19:35:26 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_fourcc.h:
	  qtdemux: make files from Vivotek camera play
	  Skip tracks of 'vivo' subtype with empty stsd instead of
	  erroring out saying that the file is broken.
	  https://bugzilla.gnome.org/show_bug.cgi?id=699791

2013-07-19 17:14:06 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>

	* gst/isomp4/gstqtmux.c:
	  qtmux: when streaming don't try to seek when stopping
	  It might cause errors in sinks that are not seekable and
	  have reported this (like e.g. fdsink)
	  https://bugzilla.gnome.org/show_bug.cgi?id=696228

2013-07-19 17:26:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: simplify some helpers
	  Some helper functions are not needed anymore or can be simplified.

2013-07-19 17:12:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: for non-raw video, move palette in caps
	  We only need to append the palette to raw video buffers, non-raw video has the
	  palette in the caps still.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-19 01:49:20 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: nitpicking in esds parsing

2013-07-19 01:49:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: set proper caps for mpeg-1 audio
	  Remove AAC specific fields from mpeg-1 audio caps, remove assumption
	  that the mpeg1 audio layer is 3, and set `parsed' field.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704548

2013-06-17 21:27:37 +0200  Arnaud Vrac <avrac@freebox.fr>

	* ext/vpx/gstvp8dec.h:
	* ext/vpx/gstvp8enc.h:
	* ext/vpx/gstvp9dec.h:
	* ext/vpx/gstvp9enc.h:
	  vpx: fix compilation when encoder or decoder headers are not installed
	  https://bugzilla.gnome.org/show_bug.cgi?id=704547

2013-07-16 20:41:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/videocrop.c:
	  videocrop: Fix unit for GRAY16 formats

2013-07-16 22:17:17 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: remove chapter stream
	  Remove all streams that are actually table of contents, since we will
	  never need the data after parsing them.

2013-07-16 21:59:37 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: send gap event for sparse streams in push mode
	  This allows to pre-roll at least if the next subtitle buffer
	  is far away.

2013-07-16 21:56:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: do not use indexes from sparse stream when seeking in push mode
	  This makes seeking more accurate in push mode, since the previous
	  keyframe on a sparse stream might be far away.

2013-07-16 21:04:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: advertise subtitle streams as sparse

2013-07-17 17:11:44 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/matroska/matroska-demux.c:
	  mastrokademux: do not push discont buffers if they aren't discont
	  Unset the discont flag instead of posssibly pushing a buffer with
	  a flag that's still set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=682110

2013-07-17 15:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: extract the palette from stsd
	  Sometimes a palette is inside the stsd, extract it instead of always using
	  the default one

2013-07-17 14:30:16 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/goom2k1/gstgoom.c:
	  goom2k1: Fix event handling and negotiate as soon as possible

2013-07-17 14:27:57 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/goom/gstgoom.c:
	  goom: Fix event handling and negotiate as soon as possible

2013-07-11 19:45:17 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: warn about the future deprecation of the "embed" property

2013-07-17 09:56:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: add support for WRAW
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-17 09:54:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: palette is appended to buffers, not in caps
	  Fix the palette handling, in 1.0 we append the palette to the buffer instead of
	  placing it on the caps.
	  See also https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-16 15:37:49 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpmp2tpay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtpmpvpay.c:
	  rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders

2013-07-15 16:24:07 +0200  Arnaud Vrac <avrac@freebox.fr>

	* gst/isomp4/qtdemux.c:
	  qtdemux: reset segment on flush stop
	  cca2f555d14 introduces a regression, where the demux segment is not
	  reset on flush stop, so the next upstream segment event will calculate
	  an invalid base time on the new segment to be sent downstream.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704255

2013-07-06 17:20:49 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: offset samples according to edit list
	  https://bugzilla.gnome.org/show_bug.cgi?id=700264

2013-07-14 12:50:13 +1200  Douglas Bagnall <douglas@halo.gen.nz>

	* tests/examples/spectrum/spectrum-example.c:
	  level: Fix the spectrum example for 1.0
	  The "message" property has been replaced by "post-messages".
	  Pre-patch output:
	  (test_spectrum:23101): GLib-GObject-WARNING **: g_object_set_valist:
	  object class `GstSpectrum' has no property named `message'
	  New spectrum message, endtime 0:00:00.100000000
	  (test_spectrum:23101): GStreamer-CRITICAL **:
	  gst_value_list_get_value: assertion `GST_VALUE_HOLDS_LIST (value)' failed
	  [...]
	  Post-patch:
	  New spectrum message, endtime 0:00:00.100000000
	  band 0 (freq 400): magnitude -65.988777 dB phase 1.533397
	  band 1 (freq 1200): magnitude -65.545563 dB phase -0.780900
	  band 2 (freq 2000): magnitude -64.791946 dB phase -0.799611
	  band 3 (freq 2800): magnitude -64.556175 dB phase -0.063615
	  [...]
	  https://bugzilla.gnome.org/show_bug.cgi?id=704179

2013-07-13 20:56:26 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: be less verbose when parsing LOAS streams
	  https://bugzilla.gnome.org/show_bug.cgi?id=704162

2013-07-12 12:31:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulsesink.h:
	  sink: alaw/mulaw caps don't have a layout property

2013-07-12 12:27:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* ext/pulse/pulseutil.c:
	  pulse: relax mulaw and alaw format checks
	  The audio library considers them as encoded formats and does not fill in the
	  sample width. The audio ringbuffers identifies the format as alaw/mulaw and that
	  is always 8 bits.

2013-07-11 16:13:05 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	* gst/isomp4/qtdemux_fourcc.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: unselect instead of ignoring disabled track, detect chapter track
	  https://bugzilla.gnome.org/show_bug.cgi?id=704007

2013-07-11 20:41:23 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: ignore errors from HEAD request
	  HEAD requests are used to check the server headers to see if it
	  seekable. Ignore errors from those requests as they shouldn't be
	  critical.
	  https://bugzilla.gnome.org/show_bug.cgi?id=704053

2013-07-12 03:24:08 +0800  Kyosuke Nekomura <supercatexpert@gmail.com>

	* gst/audiofx/audioecho.c:
	  audioecho: Fix handling of delay property in PLAYING/PAUSED state
	  https://bugzilla.gnome.org/show_bug.cgi?id=703901

2013-07-09 17:56:57 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Enable proxy caps on the src pads

2013-07-11 16:57:15 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development

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=== release 1.1.2 ===

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2013-07-11 15:58:51 +0200  Sebastian Dröge <slomo@circular-chaos.org>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
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	* configure.ac:
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	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-monoscope.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* win32/common/config.h:
	  Release 1.1.2

2013-07-11 15:58:29 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
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2013-07-09 15:34:04 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: defer the window handle setup to the main thread

2013-07-09 15:33:18 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: default to the main in case we are not setup yet

2013-07-07 22:16:05 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: close the internal window correctly

2013-07-07 21:14:22 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: only create the NS app thread for Cocoa once
	  The helper thread for Cocoa, in case no NS run loop is running,
	  should be started only once and shared across all the instances
	  running

2013-07-09 19:10:17 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: correct argument order in gst_util_uint64_scale_int_round
	  https://bugzilla.gnome.org/show_bug.cgi?id=703350

2013-07-09 17:42:59 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpmux.c:
	  rtpmux: Keep caps order from the peer or the filter

2013-07-09 12:42:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixer2.c:
	  videomixer: Fix handling of buffers without a duration
	  We'll have to pop buffer from collectpads and store it
	  internally only to get the timestamp of the next buffer.
	  If we continue to keep it in collectpads, no new buffer
	  to calculate the end time will ever arrive.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703743

2013-07-09 11:53:07 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixer2.c:
	  videomixer: Fix negotiation with 0/1 framerates
	  https://bugzilla.gnome.org/show_bug.cgi?id=703743

2013-07-09 11:17:59 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Unlock stream lock after use
	  Stream lock of sink pad was not unlocked after non-updating seek.

2013-06-27 13:26:31 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/multipart/multipartmux.c:
	  multipartmux: Re-set need_segment flag after FLUSH_STOP
	  https://bugzilla.gnome.org/show_bug.cgi?id=703182

2013-07-05 11:51:04 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: bufferpool: don't forget to release buffer on error
	  If the pool is stopped while gst_v4l2_buffer_pool_dqbuf() waits for a
	  buffer then the return value is GST_FLOW_FLUSHING. In this case the buffer
	  to queue must also be released. Otherwise is will never be deleted or
	  returned to its pool.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703764

2013-07-08 14:15:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* tests/check/elements/rtp-payloading.c:
	  rtp: Fail payloading unit test if an error message is received

2013-07-08 14:09:37 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtp/gstrtph263ppay.c:
	  rtph263ppay: Don't pass upstream filter caps to downstream
	  Downstream usually can't accept video/x-h263 but only application/x-rtp,
	  so we would always get an empty intersection here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702632

2013-07-05 22:00:37 +0200  Piotr Drąg <piotrdrag@gmail.com>

	* po/POTFILES.in:
	  po: update POTFILES.in
	  https://bugzilla.gnome.org/show_bug.cgi?id=703685

2013-07-02 11:13:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid some strdup

2013-07-02 10:37:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add select-stream signal
	  Add a signal to let the app select what streams will be selected.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=634419

2013-07-02 10:37:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: avoid strdup

2013-07-02 10:12:17 +0200  J. Rick Ramstetter <rick.ramstetter@gmail.com>

	* gst/rtp/README:
	* gst/rtpmanager/gstrtpbin.c:
	  rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin
	  https://bugzilla.gnome.org/show_bug.cgi?id=703426

2013-07-01 16:55:01 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: don't extract data from caps twice
	  gst_video_info_from_caps() always extract width, height, interlace mode and
	  framerate now. It is no longer necessary to do it again for encoded
	  formats.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703399

2013-06-20 09:41:48 -0300  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: also consider stop positions in seeks
	  Use seek stop position as range end for requests
	  https://bugzilla.gnome.org/show_bug.cgi?id=702206

2013-06-19 14:06:40 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: allow seeks in ready
	  On is_seekable, check if the server's headers have already been
	  received. If not, do a HEAD request to get them before responding
	  to basesrc.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702206

2013-07-01 17:28:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add signal to notify of the SDP
	  This way, the app can look and modify the SDP.

2013-06-21 18:10:28 +0200  Kishore Arepalli <kishore.arepalli@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: Allow negative offsets to specify offset from bottom/right
	  https://bugzilla.gnome.org/show_bug.cgi?id=702826

2013-06-30 21:01:20 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/Makefile.am:
	* gst/isomp4/qtdemux.c:
	  qtdemux: compute framerate from average sample duration
	  https://bugzilla.gnome.org/show_bug.cgi?id=703350

2013-06-25 21:16:38 +0200  Alban Browaeys <prahal@yahoo.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Add flvversion 1 to the flash-video caps
	  This allows using avdec_flv which requires this field to be
	  present in the caps. FLV only supports flash-video version 1
	  right now.
	  https://bugzilla.gnome.org/show_bug.cgi?id=703076

2013-07-01 11:37:00 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/interleave/deinterleave.c:
	  deinterleave: Don't hold object lock while sending events downstream
	  Based on a patch by Kishore Arepalli <kishore.arepalli@gmail.com>
	  https://bugzilla.gnome.org/show_bug.cgi?id=703114

2013-07-01 10:59:07 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Add MPEG4 video profile/level to the caps

2013-07-01 10:56:28 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Add AAC profile/level to the caps
	  https://bugzilla.gnome.org/show_bug.cgi?id=703312

2013-06-28 15:21:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	* gst/rtp/gstrtpvorbispay.h:
	  vorbispay: add support for config-interval
	  Align code with the theora payloader and add support for the config-interval to
	  periodically send out the config headers.

2013-06-28 15:21:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  theorapay: small cleanups

2013-06-28 12:08:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtptheorapay.c:
	  theorapay: handle streamheaders as well

2013-06-28 12:06:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	  vorbispay: always collect headers on data
	  When we see a data packet, always check if we need to collect any previous
	  headers.

2013-06-28 11:43:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpvorbispay.c:
	  vorbispay: handle streamheader as well
	  Take config strings from the streamheader when we can
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=664312

2013-06-27 07:40:29 +0200  David Svensson Fors <davidsf@axis.com>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: avoid double buffer unmap on error
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171

2013-06-27 17:02:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: reset-sync before play
	  Call reset-sync on the rtpbin before we go to playing. This makes us require SR
	  packets for all streams again before we attempt to sync them. If we don't reset,
	  it might be that we combine SR packets from before and after the PAUSE/PLAYING
	  state change and end up with huge bogus offsets.

2013-06-27 16:23:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve sync on first packets
	  Don't throw away the first RTCP packet if it arrives before the first
	  RTP packet but remember and use it to signal sync once we get the
	  RTP packet.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=691400

2013-06-27 16:15:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: only signal loop when active
	  Only signal the loop function when it is active.

2013-06-27 16:13:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: signal timestamp discont
	  We can now use the RESYNC buffer flag to mark a timestamp discont when we update
	  the ts-offset property.

2013-06-26 20:49:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegpay.c:
	  jpegpay: turn some errors into warnings
	  Turn some errors into warnings, we can continue processing so this should
	  not be fatal.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=657079

2013-06-26 14:58:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: avoid some flushes

2013-06-26 14:41:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle data message when waiting for reply
	  When we are waiting for a server reply, handle data messages instead of
	  ignoring them.

2013-06-26 14:27:34 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: handle data messages in separate method
	  Refactor and make a method to handle a data message.

2013-06-25 20:36:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: add some more docs to handle-request signal
	  See https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-10 17:20:30 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  Send a clock_provide message on the bus when we get a netclock

2013-06-10 17:20:14 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Expose use-pipeline-clock property

2013-06-24 17:11:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/udp/gstmultiudpsink.c:
	  udpsink: bind to the given interface
	  Actually call BINDTODEVICE to bind to the interface as given by the
	  property.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702819

2013-06-22 10:59:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/vpx/gstvp8dec.c:
	  vp8dec: Error out gracefully if we get an unsupported color format
	  In theory we can only get I420 though, just to be on the safe side.

2013-06-22 10:57:41 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9enc.c:
	  vp9: Add support for YV12, Y42B and Y444 color formats
	  The encoder does not work with Y42B and Y444 yet it seems.

2013-06-22 10:26:18 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/vpx/gstvp9dec.c:
	  vp9dec: Update default postproc settings from vp9_dx_iface.c

2013-06-21 13:11:32 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/webm-mux.c:
	  matroska: Add initial VP9 support

2013-06-21 13:07:30 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	* ext/vpx/Makefile.am:
	* ext/vpx/gstvp9dec.c:
	* ext/vpx/gstvp9dec.h:
	* ext/vpx/gstvp9enc.c:
	* ext/vpx/gstvp9enc.h:
	* ext/vpx/plugin.c:
	  vpx: Add initial, experimental VP9 support

2013-06-21 10:32:30 +0200  Youness Alaoui <youness.alaoui at collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtsp: go back into the loop after doing pause
	  After we do a pause request, go back to loop mode so that we can listen
	  for server messages again.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-20 23:16:17 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtpmanager/gstrtpptdemux.c:
	  rtpptdemux: Wait after the caps to forward the other events
	  First forward the stream-start, then the caps, then the rest

2013-06-21 00:42:02 +0100  Tim-Philipp Müller <tim@centricular.net>

	* sys/ximage/gstximagesrc.c:
	  ximagesrc: clear dts on buffer acquired from pool
	  When setting timestamps on outgoing buffers, clear the
	  dts explicitly, otherwise it may end up being set to a
	  bogus value from last time it was used. Avoids every
	  second or so buffer's dts being set to 0. Not that it
	  should matter for raw video.

2013-06-20 15:35:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* sys/v4l2/gstv4l2.c:
	  v4l2: don't redefine the PERFORMANCE debug variable
	  It is already defined in core.
	  fixes https://bugzilla.gnome.org/show_bug.cgi?id=702732

2013-06-20 14:43:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix race in state change to paused
	  When we go to paused, we first flush the connection and then send the pause
	  command. As a result of the flushing, the scheduled paused command can get
	  lost. Wait until the connection is completely flushed and the rtsp task is
	  waiting before issuing the paused or playing request.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-20 11:31:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: handle SEGMENT query

2013-06-19 12:37:31 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* sys/v4l2/gstv4l2src.c:
	  v4l2: Optimize negotiation by removing the query filter
	  As cameras tend to have a quite specific set of capabilities (specific
	  framerates for each resolution), getting the peer caps filtered by our
	  probed caps can cause a big increase in the caps size which slows down
	  things quire a bit.
	  As for negotiation v4l2 iterates through the caps of the peer to find the
	  first intersection with the probed caps, getting the fully expanded
	  intersection of capabilities is not useful.
	  Using the same testcase as for bug #702632, adding this patch on top of
	  the patches suggested there speeds up getting the inital frame from
	  around ~14-15 seconds to around ~3-4 seconds.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702638

2013-06-19 10:30:56 +0200  Kishore Arepalli <kishore.arepalli@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: duration query returns zero for DV video in avi
	  https://bugzilla.gnome.org/show_bug.cgi?id=702625

2013-06-19 11:06:37 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Disable usage of allocation queries
	  This can only reliably work if demuxers have a
	  separate streaming thread per srcpad. This should be
	  done in a demuxer base class, which integrates parts
	  of multiqueue
	  https://bugzilla.gnome.org/show_bug.cgi?id=701856

2013-06-11 15:02:21 +0100  Alex Ashley <bugzilla@ashley-family.net>

	* gst/isomp4/qtdemux.c:
	  Avoid skipping moov atoms for fragmented MP4 files.
	  bug #700505
	  Following a representation change that causes a resolution change,
	  the video decoder fails to decode correctly. Dashdemux detects the
	  representation change and pushes a new caps event and an
	  initialization segment (a new moov atom) to the downstream qtdemux,
	  but it doesn't handle this new moov yet, it will only parse the
	  first one it receives.
	  This commit changes qtdemux to accept a new moov in a dash bitstream
	  switching scenario.

2013-06-19 00:42:54 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: send stream-start only once for each stream
	  Do not send stream start again when reconfiguring a pad for new caps.
	  That is common for adaptive streams

2013-06-05 17:02:49 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* sys/osxvideo/cocoawindow.m:
	* sys/osxvideo/osxvideosink.m:
	  osxvideosink: fix support in VM's without hardware acceleration

2013-06-15 12:29:31 +0200  Jens Georg <mail@jensge.org>

	* gst/rtp/gstrtpmp2tdepay.c:
	  rtpmp2tdepay: accept mislabelled streams from GStreamer 0.10 as well
	  The mp2t payloader in 0.10 mislabelled the streams as MP2T-ES
	  instead of MP2T, so accept that as well for compatibility reasons.
	  https://bugzilla.gnome.org/show_bug.cgi?id=702457

2013-06-16 05:40:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: manage element state ourselves
	  Lock the state of the all our elements and manage their states
	  outselves. Because we are working async, we can't rely on the state
	  change function to set the state at the right time or to return the
	  right return value from the state change function.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046

2013-06-14 14:09:50 +0200  Bruno Gonzalez <stenyak@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't unlock stream lock without locking it first
	  https://bugzilla.gnome.org/show_bug.cgi?id=702167

2013-06-13 16:00:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Use the right hashtable to calculate bandwidth
	  Don't use an unused hashtable to iterate source to calculate bandwidth.
	  Remove unused code.

2013-06-12 16:27:24 -0600  Brendan Long <b.long@cablelabs.com>

	* configure.ac:
	  pulsesink: Require PulseAudio >= 2.0
	  This is needed for pa_format_info_get_prop_* functions.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-13 14:23:08 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	  Revert "pulsesink: Make 2.0 dependency optional"
	  This reverts commit 01457027e0d384aca3e551ae684e0aa074ee5498.
	  We'll just depend on PulseAudio 2.0 or above instead of having the bug
	  partially fixed based on the installed libpulse version.

2013-06-13 12:40:15 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* configure.ac:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulseutil.c:
	  pulsesink: Make 2.0 dependency optional
	  The getcaps function we added uses some pa_format_info_get_prop...
	  accessor functions that were only added in 2.0, so we only have our
	  getcaps implementation exist if we're compiling against libpulse 2.0 or
	  above.
	  Eventually, we could bump the minimum requirement to 2.0 or above.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-12 18:23:46 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/videomixer/videomixer2.c:
	  Revert "videomixer: When all sinkpads are eos, update output segment stop and forward it"
	  This reverts commit 2d3910fc7901b5f29e16c0fdd4e9067a6d7f66fe.
	  It's not solving any problem and instead causes code to fall apart.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701519

2013-01-09 09:39:33 +0000  Tim-Philipp Müller <tim@centricular.net>

	* gst/matroska/matroska-demux.c:
	  matroskademux: mark subtitle streams as sparse in stream-start event
	  And also mark the streams that should be selected by default if
	  marked so in the headers.
	  https://bugzilla.gnome.org/show_bug.cgi?id=600648

2013-06-11 22:12:58 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanoramaorc-dist.c:
	* gst/audiofx/audiopanoramaorc-dist.h:
	  audiopanorama: add prebuilt files

2013-06-11 20:27:51 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c:
	  audiopanorama: cleanup and expand the tests
	  Split out two more tests. Extract more common code into helpers. Add coverage for float.

2013-06-10 21:15:20 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	  audiopanorama: cleanup of transform()
	  Only map input if we are reading it. Cleanup the logging and the comments a bit.

2013-06-09 20:35:18 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiopanorama.h:
	* gst/audiofx/audiopanoramaorc.orc:
	  audiopanorama: use orc to speedup processing
	  Use special variants for the case when we don't change the panorama (pan=0.0).
	  Simplify the processing functions by passing the panorama value directy instead
	  of the instance. Use orc for clearing buffers too.

2013-06-11 19:24:49 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: check last end_time after conversion to running segment
	  The last end_time was saved after conversion, so the comparison
	  had to be made after conversion for it to make sense.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701385

2013-06-11 19:22:20 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: add mix->segment.start to output_end_time
	  When the segment start is not 0, this created a situation where
	  the output_end_time is inferior to output_start_time, and the duration
	  of the next buffer ended up underflowing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701385

2013-06-11 13:54:53 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Send stream headers after the segment event
	  https://bugzilla.gnome.org/show_bug.cgi?id=700799

2013-06-11 12:26:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Do allocation query after exposing all pads and no-more-pads
	  Also configure video streams as early as possible.
	  Related https://bugzilla.gnome.org/show_bug.cgi?id=701856
	  but not fixing that.

2013-06-11 12:25:46 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Don't forward CAPS events from upstream
	  Just use the default pad event handler.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701976

2013-05-26 08:18:04 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Cache the getcaps/acceptcaps probe stream
	  getcaps is called frequently during stream setup, and creating a new
	  stream each time is very inefficient. There's some more room for
	  optimisation by caching the queried sink formats as well, but this needs
	  some more changes to listen for format changes on the sink (for when
	  supported formats change between probe stream creation and sink
	  querying).
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-05-23 21:39:08 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesink: Add a getcaps function
	  This allows us to have more fine-tuned caps in READY or above. However,
	  this is _really_ inefficient since we create a new stream and query sink
	  for every getcaps in READY, which on a simple gst-launch line happens
	  about 35 times. The next step is to cache getcaps results.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-05-10 11:32:44 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	  pulsesink: Take a lock on the ringbuffer in acceptcaps
	  This is needed as a concurrent state change could pull the context or
	  stream out from under our feet.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-09 20:29:09 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/audiofx/audiopanorama.c:
	* gst/audiofx/audiopanorama.h:
	  audiopanorama: move the enum to the header and use instead of gint
	  Move the enum for the processing method to the header so that we can use the
	  type for the instance struct.

2013-06-09 20:32:22 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/level.c:
	  level: rework the tests to cover other formats too

2013-06-05 16:32:30 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2: make sure the element is not deleted before the pool
	  The pool accesses data from the v4l2object so it must exist at least
	  as long as the pool. Refcount the element which controls the object
	  live-time.
	  https://bugzilla.gnome.org/show_bug.cgi?id=701650

2013-06-07 15:38:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* ext/libpng/Makefile.am:
	  png: Link with libgstbase for GstByteReader and GstAdapter

2013-06-07 15:15:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavenc/Makefile.am:
	  wavenc: Link with libgstbase for GstByteWriter

2013-06-07 13:26:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Push stream-start event in pull mode before anything else

2013-05-10 12:09:19 +0530  Arun Raghavan <arun.raghavan@collabora.co.uk>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Get rid of acceptcaps side-effects
	  The sink info callback should not have side-effects on the GstPulseSink
	  object since we are sometimes using with a dummy stream in acceptcaps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=686459

2013-06-05 18:36:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  Back to development

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=== release 1.1.1 ===

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2013-06-05 17:58:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
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	* ChangeLog:
	* NEWS:
	* RELEASE:
	* common:
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	* configure.ac:
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	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.interfaces:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-deinterlace.xml:
	* docs/plugins/inspect/plugin-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multifile.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-soup.xml:
	* docs/plugins/inspect/plugin-spectrum.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-video4linux2.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	* docs/plugins/inspect/plugin-y4menc.xml:
	* gst-plugins-good.doap:
	* gst/deinterlace/tvtime-dist.c:
	* gst/deinterlace/tvtime-dist.h:
	* gst/videobox/gstvideoboxorc-dist.c:
	* gst/videobox/gstvideoboxorc-dist.h:
	* gst/videomixer/blendorc-dist.c:
	* gst/videomixer/blendorc-dist.h:
	* win32/common/config.h:
	  Release 1.1.1

2013-06-05 16:35:19 +0200  Sebastian Dröge <sebastian.droege@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/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files
Sebastian Dröge's avatar
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