Release 1.3.11.3.1

81 files changed, 6563 insertions, 319 deletions

diff --git a/ChangeLog b/ChangeLog
index 36f60b8a3..136ab1035 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,5269 @@
+=== release 1.3.1 ===
+
+2014-05-03  Sebastian Dröge <slomo@coaxion.net>
+
+	* configure.ac:
+	  releasing 1.3.1
+
+2014-05-03 17:22:45 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* po/da.po:
+	* po/de.po:
+	* po/el.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/lv.po:
+	* po/nb.po:
+	* po/pt_BR.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sl.po:
+	* po/sr.po:
+	* po/zh_CN.po:
+	  po: Update translations
+
+2014-05-03 11:43:21 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/shapewipe.c:
+	  shapewipe: Send initial events after setting the elements to PLAYING
+	  Otherwise we send them too early, and setting the elements to PLAYING
+	  afterwards will drop all the events again.
+
+2014-05-03 10:15:03 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From bcb1518 to 211fa5f
+
+2014-05-02 17:12:29 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: Set segment position to the stop position of the buffer
+
+2014-05-02 17:10:18 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: Properly report errors before stopping the srcpad task
+
+2014-05-02 17:02:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: Error out if we have no caps yet
+
+2014-05-02 14:49:27 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: avoid dividing by a 0 blockalign
+	  This can be 0. In that case, do not try to cut off the last few
+	  bytes from the last buffer.
+	  Coverity 1146971
+
+2014-05-02 14:25:01 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: do not use uinitialized clut on error
+	  If we're missing part of the clut, do not try to use it. It seems
+	  very likely the break was meant to break out of the switch rather
+	  than from the loop.
+	  Coverity 1139878
+
+2014-05-02 14:18:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/flx/gstflxdec.c:
+	  flxdec: fix integer overflow
+	  Coverity 1139859
+
+2014-05-02 14:09:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtp/gstrtpqdmdepay.c:
+	  rtpqdmdepay: remove pointless check
+	  Besides, the pointer was dereferenced earlier anyway.
+	  Coverity 1139853
+
+2014-05-02 14:06:25 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: remove duplicate test
+	  item was dereference previously.
+	  While there, reorder some test for faster early out.
+	  Coverity 1139844
+
+2014-05-02 14:02:52 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/vpx/gstvp8enc.c:
+	  vp8enc: guard against NULL pointer dereference
+	  Coverity 1139838
+
+2014-05-02 13:59:07 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: fix theoretical integer overflow
+	  This code isn't actually used at the moment, unsure if I should
+	  just remove it or not...
+	  Coverity 1139811
+
+2014-05-02 13:33:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/ebml-write.c:
+	  matroska: blindly fix writing variable length negative values
+	  Spotted while fixing something else in the area.
+	  Nothing calls this with a negative value.
+
+2014-05-02 13:29:33 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/ebml-write.c:
+	  matroska: do not lose the top bits when writing a > 32 bit value
+	  Coverity 1139806
+
+2014-05-02 12:10:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/videofilter/gstvideoflip.c:
+	  videoflip: add missing break in switch
+	  Coverity 1139755
+
+2014-05-02 11:39:39 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-parse.c:
+	  matroska: do not try to call gst_pad_query_default on a NULL pad
+	  gst_matroska_parse_query can be called explicitely with a NULL pad.
+	  If we reach this point with a NULL pad, fail the query.
+	  Coverity 1139715
+
+2014-05-02 11:28:01 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-parse.c:
+	  matroska: do not return GST_FLOW_OK if we did not get a buffer
+	  Coverity 1139714 (which will likely come back in another guise,
+	  as the _read_init call can have a failing _map)
+
+2014-05-02 11:20:33 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/ebml-write.c:
+	  matroska: catch failure to map buffer
+	  Avoids dereferencing NULL.
+	  Coverity 1139712
+
+2014-05-02 10:52:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/avi/gstavimux.c:
+	  avimux: refuse caps with invalid framerate
+	  Coverity 1139701
+
+2014-05-02 10:21:09 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: handle 0 size packets without dividing by 0
+	  Coverity 1139691
+
+2014-05-02 09:49:32 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: guard against invalid frame size to avoid division by 0
+	  Coverity 1139690
+
+2014-05-02 09:49:17 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: trivial typo fix
+
+2014-05-02 09:43:54 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/speex/gstspeexdec.c:
+	  speexdec: remove dead code
+	  fpp can never equal 0 here, or the loop would not execute at all.
+	  Zero fpp was possible before as the loop condition was allowing
+	  it specifically, but no more.
+	  Coverity 1139681
+
+2014-05-02 09:41:19 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* sys/oss4/oss4-property-probe.c:
+	  oss4: remove dead mixer code
+	  This was partly removed in the port to 0.11. If still needed,
+	  it's still there in the history.
+	  Coverity 1139687
+
+2014-05-02 09:33:51 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* sys/oss4/oss4-property-probe.c:
+	  oss4: fix a missing unlock and a return-only-when-assertions-enabled
+	  Spotted on the side while looking at another issue.
+
+2014-03-07 17:31:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Correctly map RGB32 format
+	  In v4l2 specification, RGB32 has the alpha, or pading, first, not last.
+	  See http://linuxtv.org/downloads/v4l-dvb-apis/packed-rgb.html .
+	  https://bugzilla.gnome.org/show_bug.cgi?id=540941
+
+2014-04-30 18:06:40 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: remove dead code
+	  For 8 bit width, we always have depth==gdepth==width==8.
+	  Coverity 1139678
+
+2014-04-30 17:48:53 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstmpegaudioparse.c:
+	  mpegaudioparse: remove dead code
+	  A stricer check is already done earlier, and integer overflows
+	  do not seem possible here.
+	  Coverity 1139675
+
+2014-04-30 14:50:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtp/gstrtpvrawpay.c:
+	  rtpvrawpay: guard against pathological "no space" condition
+	  Even if one woul hope one pixel can fit in a MTU, ensure we do not
+	  overwrite a buffer if this is not the case.
+	  Spotted while looking at Coverity 1208786
+
+2014-04-30 11:52:10 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtp/gstrtpjpegdepay.c:
+	  rtpjpegdepay: sanity check for NULL qtable
+	  Can happen (at least in crafted stream)
+	  Coverity 1208778
+
+2014-04-30 01:08:41 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: pass on tags from upstream if there are any
+	  Don't just ignore upstream tags from e.g. an ID3 tag before
+	  the .wav data, pass them on downstream.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=729223
+
+2014-04-29 16:26:53 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: optimize timer update
+	  When we are not doing retransmission, we just need to find the current
+	  seqnum so we can stop when we found it.
+
+2014-04-29 16:21:44 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.h:
+	  rtpjitterbuffer: small optimizations
+	  Small optimizations where we can.
+	  Add some more debug.
+
+2014-04-29 16:16:17 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: signal when next_seqnum changed
+	  Signal the pushing thread when the next_seqnum changed and we might be
+	  able to push a buffer now.
+
+2014-04-29 16:12:29 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: only signal event when head changed
+	  After adding a buffer, only signal the pushing thread when the head
+	  buffer changed or else we cause a useless wakeup.
+
+2014-04-29 15:29:31 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  rtpjitterbuffer: rework packet insert
+	  Rework the packet queue so that the most common action (insert a packet
+	  at the tail of the queue) goes very fast.
+	  Report if a packet was inserted at the head instead of the tail so that
+	  we can know when to retry _pop or _peek.
+
+2014-04-28 14:41:10 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtp/gstrtpvrawdepay.c:
+	* gst/rtp/gstrtpvrawpay.c:
+	  rtpvraw: use plane pointers when needed
+	  Pack/unpack planar formats to/from the first plane.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729058
+
+2014-04-28 09:47:10 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: Remember if a redirect is permanent or not and store it in the query
+
+2014-04-27 21:57:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* gst/goom/config_param.c:
+	  goom: Remove french comment saying to prefix functions
+	  All non-static function in this file are already prefixed with goom_.
+
+2014-04-28 00:20:47 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/goom/filters.c:
+	  goom: fix compilation on ios-arm7-10.9 and osx-x86_64
+	  uint is not a standard type, and the rest of the code uses
+	  Uint which is locally typedefed to unsigned int.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=729067
+
+2014-04-27 18:29:11 -0400  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/goom/filters.c:
+	  goom: fix undefined behaviour of left-shift
+	  Don't left-shift into the sign bit, the result is undefined and potentially
+	  an overflow could flip the sign.
+
+2014-04-26 20:51:36 -0400  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: check return from qt_demux_video_caps
+	  Now qtdemux_video_caps() can return NULL. We need to check this return before
+	  using it's value.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=728987
+
+2014-04-26 23:35:17 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/dv/gstdvdec.c:
+	* ext/dv/gstdvdemux.c:
+	* ext/jack/gstjackaudiosink.c:
+	* ext/jack/gstjackaudiosrc.c:
+	* ext/speex/gstspeexdec.c:
+	* gst/avi/gstavidemux.c:
+	* gst/avi/gstavisubtitle.c:
+	* gst/isomp4/gstqtmoovrecover.c:
+	* gst/isomp4/gstqtmux-doc.c:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/qtdemux.c:
+	* gst/multifile/gstmultifilesink.c:
+	* gst/rtp/gstrtpL16depay.c:
+	* gst/rtp/gstrtpL16pay.c:
+	* gst/rtp/gstrtpac3depay.c:
+	* gst/rtp/gstrtpac3pay.c:
+	* gst/rtp/gstrtpamrdepay.c:
+	* gst/rtp/gstrtpamrpay.c:
+	* gst/rtp/gstrtpbvdepay.c:
+	* gst/rtp/gstrtpbvpay.c:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpmux.c:
+	* gst/rtpmanager/gstrtpptdemux.c:
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	* gst/rtsp/gstrtpdec.c:
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/spectrum/gstspectrum.c:
+	* gst/udp/gstudpsrc.c:
+	* gst/videofilter/gstgamma.c:
+	* gst/videofilter/gstvideobalance.c:
+	* gst/videofilter/gstvideoflip.c:
+	* gst/wavparse/gstwavparse.c:
+	* sys/osxaudio/gstosxaudiosink.c:
+	  docs: remove outdated and pointless 'Last reviewed' lines from docs
+	  They are very confusing for people, and more often than not
+	  also just not very accurate. Seeing 'last reviewed: 2005' in
+	  your docs is not very confidence-inspiring. Let's just remove
+	  those comments.
+
+2014-04-25 17:58:42 -0400  Luis de Bethencourt <luis@debethencourt.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: initialize caps pointer to null
+	  Make sure the caps pointer returns initialized when using it in
+	  qtdemux_parse_tree ().
+	  https://bugzilla.gnome.org/show_bug.cgi?id=728987
+
+2014-04-22 17:07:38 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Clear last_pt on flush-stop.
+	  Otherwise, we don't recheck the buffer caps for clock-rate
+	  properly on the next chain.
+
+2014-04-22 17:29:02 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Fix compiler warning
+	  gstdeinterlace.c: In function 'gst_deinterlace_output_frame':
+	  gstdeinterlace.c:1537:57: error: 'pattern.length' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+	  This actually is always initialized before it is used there, but
+	  let's just silence gcc here.
+
+2014-04-21 15:58:45 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpmux.c:
+	  rtpmux: fix buffer list drop check
+	  While porting to 0.11, the check was mistakenly made constant,
+	  instead of testing for the return value of process_buffer_locked.
+	  Coverity 1139663
+
+2014-04-21 13:44:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroska: fix content encoding scope validity check
+	  It's 3 bits, and http://matroska.org/technical/specs/index.html
+	  says it can't be 0.
+	  Coverity 1139660
+
+2014-04-21 13:34:37 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: fix PAR fraction sanity check
+	  It was checking par_num twice, and never par_denum.
+	  Coverity 1139634
+
+2014-04-21 13:32:40 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiidpsink: warn when setsockopt fails
+	  This doesn't seem to be fatal, but it's good to let the user know
+	  in the logs.
+	  Coverity 1139630
+
+2014-04-21 13:27:24 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/interleave/deinterleave.c:
+	  interlace: catch failure to create audio info from caps
+	  Coverity 1139627, 1139628
+
+2014-03-13 09:37:48 +0100  Göran Jönsson <goranjn@axis.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	  gstrtph264pay: Reset sps pps variable when state change.
+	  Reset last_spspps and sps/pps arrays  when state transition
+	  GST_STATE_CHANGE_PAUSED_TO_READY.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726015
+
+2014-04-18 11:11:14 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  jitterbuffer: improve EOS handling
+	  Make a new method to disable the jitterbuffer buffering.
+	  Rework the update_estimated_eos() method. Calculate how much time
+	  there is left to play. If we have less than the delay of the
+	  jitterbuffer, we disabled buffering because we might never be able to
+	  fill the complete jitterbuffer again.
+	  If we receive an EOS event, disable buffering. We will drain the
+	  buffer and eventually push the EOS event out.
+	  When we reach the estimated NPT timeout and we didn't receive an EOS
+	  event, make one and queue it so that it can be pushed.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
+
+2014-04-18 10:21:27 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	  rtpsession: send reconfigure when internal-ssrc changes
+	  When the internal-ssrc property changes, we want to send a reconfigure
+	  upstream to make payloaders use the new suggested ssrc.
+	  Using the internal-ssrc property to change the SSRC of a stream is not a
+	  good idea and doesn't work when there are multiple senders, we want to
+	  set the SSRC directly on the payloaders. Therefore, deprecate this
+	  property.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725361
+
+2014-04-18 04:23:26 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: assume a full buffer when eos
+	  Rework the logic to make buffering messages a little, make sure we
+	  don't make the same message multiple times.
+	  Consider the buffer full when EOS was received.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
+
+2014-04-17 18:07:09 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/rtprtx.c:
+	  rtprtx: Don't forget to unmap rtp buffer in the test
+
+2014-04-17 17:58:58 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: Require clock-rate in the caps and handle no ssrc in the caps properly
+
+2014-04-17 17:43:12 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/rtprtx.c:
+	  rtprtx: Provide an ssrc in the test
+	  And increase timeout to allow all tests to run in valgrind.
+
+2014-04-17 17:33:46 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/rtpsession.c:
+	  rtpsession: Fix memory leaks in test
+
+2014-04-17 17:26:36 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix hundreds of memory leaks in the test
+
+2014-04-17 17:00:37 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Unref clock id when waiting for the clock is interrupted
+
+2014-04-17 16:39:59 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/rtpcollision.c:
+	  rtpcollision: Fix memory leaks in unit test
+
+2014-04-16 21:40:45 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: name collectpads object based on videomixer name
+	  Makes it easier to track things in debug logs when there
+	  are multiple mixers and muxers.
+
+2014-04-16 21:37:12 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: better logging of incoming events
+	  The pad and parent names are already logged as part of logging
+	  the object. Instead log the full event details.
+
+2014-04-16 19:03:47 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/videomixer.c:
+	  videomixer: Fix memory leak in unit test
+
+2014-04-16 18:49:43 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/level/gstlevel.c:
+	  level: Use the correct number of samples to iterate over the input array
+	  Fixes invalid memory accesses and accesses to uninitialised data.
+
+2014-04-16 18:00:49 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/icydemux/gsticydemux.c:
+	  icydemux: Unref dropped events
+
+2014-04-16 17:29:30 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/ebml-read.c:
+	  matroska: fix check for amount of data to read
+	  History shows length==0 should set data to NULL and return,
+	  so we do that too instead of trying to read nothing.
+	  Coverity 206205
+
+2014-04-16 17:25:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: fix sign comparison
+	  history_count is unsigned, so the whole comparison will be made
+	  as unsigned, and fail to reject what it was meant to.
+	  Coverity 206204
+
+2014-04-16 17:04:50 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: remove dead code
+	  sub may not be NULL in this switch, there is a bail out just
+	  before it if so.
+	  Coverity 206098
+
+2014-04-16 16:59:43 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstflacparse.c:
+	  flacparse: remove dead code
+	  The block_size == 0 was shortcut earlier, and the variable is not
+	  modified in the meantime.
+	  Coverity 206097
+
+2014-04-16 16:56:54 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/videomixer/videoconvert.c:
+	  videomixer: remove dead code
+	  While it seems to keep a compile time selection, I traced it
+	  to some code copied from videoconvert, where it was removed,
+	  with the following comment:
+	  Also remove the high-quality I420 to BGRA fast-path as it needs
+	  the same fix, which causes an additional instruction, which causes
+	  orc to emit more than 96 variables, which then just crashes.
+	  This can only be fixed in orc by breaking ABI and allowing more
+	  variables.
+	  Thus, I remove it here as well.
+	  Coverity 206064
+
+2014-04-16 16:50:30 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/qtdemux.c:
+	  isomp4: fix incorrect masking for multiple tags
+	  Coverity 206058
+
+2014-04-16 16:45:08 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/atoms.c:
+	  isomp4: fix wrong atom flags set when adding samples
+	  Coverity 206057
+
+2014-04-16 16:40:02 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audiofx/audiofxbasefirfilter.c:
+	  audiofx: fix comparison of delta time to a threshold
+	  Coverity 206055
+
+2014-04-16 16:32:26 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: do not rely on call failure keeping return data unmodified
+	  This is clearer this way too.
+	  Coverity 206029
+
+2014-04-16 16:28:49 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/atomsrecovery.c:
+	  isomp4: catch fseek error
+	  Coverity 206028
+
+2014-04-16 16:25:44 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/isomp4/atoms.c:
+	  isomp4: report failures to caller
+	  Coverity 206027
+
+2014-04-16 18:05:46 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: refuse serialied query when buffering
+	  When we are buffering, we can't block and wait for the serialized query
+	  to complete because the jitterbuffer will not try to forward the query
+	  while buffering. Instead, just refuse the query.
+
+2014-04-16 16:51:15 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: don't free the serialized query
+	  We should never free a serialized query in the queue, it is the upstream
+	  caller that will free it.
+
+2014-04-16 17:35:42 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/aacparse.c:
+	  aacparse: Fix memory leak in the test
+
+2014-04-16 17:33:46 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: Create hashtable only when we actually use it
+	  In error cases we previously returned without freeing it.
+
+2014-04-16 17:30:59 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: Chain up to the parent class' dispose function
+
+2014-04-16 17:23:27 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: Initialise ioctl struct with zeroes before passing it to ioctl()
+
+2014-04-16 13:47:43 +0200  Marc Leeman <marc.leeman@gmail.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: correct LOG msg for -1
+	  Signed-off-by: Marc Leeman <marc.leeman@gmail.com>
+
+2014-04-15 21:36:30 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/interleave/interleave.c:
+	  interleave: Fix negotiation to work at all again
+	  The caps query handling function for the sinkpads was called for
+	  the srcpad, and the sinkpads had none. This commit moves it to the
+	  right pad, but nonetheless the negotiation still looks wrong.
+	  This makes the test pass again after the recent coverity fix
+	  and also allows interleave to work again, but someone should
+	  really review the negotiation code and fix it.
+
+2014-04-13 09:03:41 +0200  Edward Hervey <edward@collabora.com>
+
+	* sys/oss4/oss4-audio.c:
+	  oss4: Maximum number of channels support is 8
+	  Avoids doing potential overwrites in ch_layout (which only has 8
+	  fields).
+	  CID #1139826
+
+2014-04-12 22:16:37 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* sys/osxvideo/osxvideosink.m:
+	  osxvideosink: Set rank to MARGINAL
+	  If available we prefer using glimagesink over osxvideosink. It supports
+	  more formats and in general has more features than osxvideosink.
+
+2014-04-11 18:19:49 +0200  Josep Torra <n770galaxy@gmail.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: only guess AU boundaries when aren't indicated by marker
+	  The marker bit isn't mandatory and we had in place code to guess AU
+	  boundaries by detecting a new picture start. This guessing code
+	  didn't work with interlaced content that has proper marker bits
+	  to indicate the AU boundaries. It was leaking the first field buffer
+	  and producing a corrupted output.
+	  fixes: https://bugzilla.gnome.org/show_bug.cgi?id=728041
+
+2014-04-10 10:38:19 -0300  Rafał Mużyło <galtgendo@o2.pl>
+
+	* ext/libpng/gstpngdec.c:
+	  pngdec: enable libpng interlaced picture handling
+	  Makes libpng deinterlace Adam7 interlaced pictures
+	  by default. It is the only interlaced format available
+	  and if the picture isn't interlaced the code should behave
+	  as before.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=726161
+
+2014-04-11 13:27:42 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Only keep-alive the connection in stop() if we have finished all previous messages
+	  After cancelling a request we need to create a new connection.
+
+2014-04-11 11:54:12 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* ext/dv/gstdvdec.c:
+	  dvdec: Don't set bogus timestamp/duration
+	  This will happen if we have an incoming stream with a non-TIME segment
+	  Could be improved later to figure out proper pts/duration.
+	  CID #1199702
+	  CID #1199703
+
+2014-04-11 11:53:42 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* ext/dv/gstdvdec.c:
+	  dvdec: Properly refuse incoming stream without framerate
+	  The return value wasn't properly propagated back if the caps
+	  didn't contain a framerate
+
+2014-04-10 16:35:28 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Also retry on unexpected network failures
+
+2014-04-10 15:45:41 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: New property to specify the maximum number of retries before we give up
+
+2014-03-13 10:56:11 +0100  Alexander Zallesov <zallesov@gmail.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Change default timeout to 15 seconds
+	  If nothing happens after 15 seconds, chances are good that
+	  our connection will never will work. Stop after 15 seconds
+	  instead of waiting until the system's default timeout, which
+	  can be > 1 minute.
+
+2014-04-09 17:30:54 +0900  Jimmy Ohn <yongjin.ohn@lge.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: replace duplicated variable when parsing trex atom
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727878
+
+2014-04-09 10:56:29 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Use GST_FLOW_FLUSHING when flushing, not GST_FLOW_EOS
+	  ... and reset it properly after flushing is done. Fixes playback
+	  in many cases when buffering is used.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727821
+
+2014-04-09 08:58:04 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Properly return stream flags when parsing trex atom
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727867
+
+2014-03-19 19:18:11 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* sys/osxvideo/osxvideosink.h:
+	* sys/osxvideo/osxvideosink.m:
+	  osxvideosink: use the video frame API instead of the video meta API
+	  https://bugzilla.gnome.org/show_bug.cgi?id=726738
+
+2014-03-19 18:47:39 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* sys/osxvideo/osxvideosink.m:
+	  osxvideosink: advertize video meta API support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=726737
+
+2014-04-08 11:31:06 +0200  Edward Hervey <edward@collabora.com>
+
+	* gst/interleave/interleave.c:
+	  interleave: Add missing break in switch statement
+	  The caps query is handled entirely already before.
+	  CID #1139757
+
+2014-04-06 18:03:11 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: souphttpsrc: use SoupKnownStatusCode if needed
+	  From libsoup docs:
+	  Prior to 2.44 SoupStatus was called SoupKnownStatusCode,
+	  but the individual values have always had the names they
+	  have now.
+	  Fixes:
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727329
+
+2014-04-07 12:58:23 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: use frames, not bytes, for position query in VBR streams
+	  Coverity 1139648
+
+2014-04-07 12:42:14 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/smpte/gstsmpte.c:
+	  smpte: fix copy/paste error causing unmap on wrong buffer
+	  Coverity 1139647
+
+2014-04-07 12:16:17 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: guard against finding no suitable pattern
+	  The code handles a -1 pattern index, and it seems plausible
+	  that a pattern might be found later, so it seems best to not
+	  send an element error here.
+	  Coverity 1139766
+
+2014-04-04 17:38:14 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: update for new MIKEY API
+
+2014-04-03 17:40:01 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: send sender SSRC in the MIKEY message
+	  Allocate a new SSRC for our RTCP messages back to the server and set
+	  this in the MIKEY message.
+
+2014-04-03 17:39:30 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: make random number for the CSB
+	  As recommended in the RFC
+
+2014-03-26 12:10:44 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: don't put spaces in keymgmt header
+
+2014-03-25 17:47:49 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: create and send the RTCP encryption key
+	  Create and make a key for encrypting the RTCP packets back to the server
+	  and wrap this in a MIKEY message that we send as a header in the SETUP
+	  request.
+
+2014-04-03 12:18:39 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: free the srtpdec element
+
+2014-04-03 12:16:25 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: cleanup stream_free function
+	  There is no reason to NULL all fields, we will free the stream anyway.
+
+2014-04-03 12:07:31 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: demote warning to debug
+	  For TCP, it is normal that we don't have timestamps so don't WARN on
+	  it.
+
+2014-03-29 19:13:06 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2: Fix support for caps without width, height, framerate or format
+	  For format like mpegts, width and height is rarely in the negotiated caps. This
+	  patch fixes failure when setting format, and prevent introducing width, height,
+	  framerate and format to the caps when fixating.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725860
+
+2014-03-31 18:34:13 +0200  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/avi/gstavidemux.c:
+	* gst/avi/gstavidemux.h:
+	  avidemux: Always set PTS=DTS on raw video streams
+
+2014-03-31 18:31:22 +0200  Thibault Saunier <tsaunier@gnome.org>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Always set pixel-aspect-ratio on raw video streams
+	  That field is mandatory in caps and if it is not present in the
+	  AVI container, it means square pixels thus 1/1.
+
+2014-03-30 00:35:07 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska-mux: add mapping for Opus audio
+	  Might want to consider adding channels/rate
+	  requirement to template caps, but requires
+	  fixing up of encoder and parser first.
+
+2014-03-30 00:31:11 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.h:
+	  matroska-demux: add mapping for Opus audio codec
+	  https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2014-03-29 17:21:17 -0400  William Manley <will@williammanley.net>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2src: Fix support for mpegts streams
+	  It seems that GStreamer's mpegts elements (tsdemux, tsparse) require caps
+	  `video/mpegts,systemstream=true`.  As far as I can see the significance
+	  of systemstream is to indicate that this is a container format rather than
+	  an elementary stream.  As this is the case (and I can't understand how it
+	  could not be the case with mpegts) I add systemstream=true to v4l2src's
+	  caps.
+	  This allows v4l2src to be linked with tsdemux for playback from my
+	  Hauppauge HD-PVR with the pipeline:
+	  v4l2src ! queue ! tsdemux ! video/x-h264 ! decodebin ! xvimagesink
+	  In combination with the next commit this fixes using Hauppauge HD-PVR with
+	  GStreamer 1.0+.
+
+2014-01-14 14:48:42 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: attempt to fix infinite (for small version of infinite) loop
+
+2014-03-29 13:20:30 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpmanager: copy sticky events when exposing pads in more places
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724712
+
+2014-03-28 20:11:36 +0100  Rico Tzschichholz <ricotz@ubuntu.com>
+
+	* sys/v4l2/Makefile.am:
+	  v4l2: fix distcheck
+	  Make sure ext/*.h are dist'ed
+
+2014-03-27 19:51:50 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/ximage/gstximagesrc.c:
+	  ximagesrc: only extrapolate alpha mask for 32-bit depth
+	  Instead of passing bogus alpha mask values when there's no alpha.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=726833
+
+2014-03-21 13:03:17 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/ximage/gstximagesrc.c:
+	  ximagesrc: Add ARGB/BGRA support
+
+2014-03-20 15:28:26 +0100  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/rtp/gstrtpjpegpay.c:
+	  jpegpay: consider header len when calculating payload len
+	  Fixed https://bugzilla.gnome.org/show_bug.cgi?id=726777
+
+2014-03-26 08:03:22 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: All frames are sync points
+
+2014-03-26 08:02:43 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/libpng/gstpngdec.c:
+	  pngdec: All frames are sync points
+
+2014-03-22 17:07:46 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: segment closing not needed in 1.x
+	  ... as sender should keep track of segment base accumulation.
+	  Rather, it may have some adverse effects as a spurious segment event,
+	  e.g. in collectpads.
+
+2014-03-22 17:05:17 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: early sending pending codec-data for all streams
+	  ... at least before syncing across all streams might cause some gap
+	  activity on any of those streams, notably sparse streams.
+	  See also #712134
+
+2014-03-22 17:01:27 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: handle both sticky and non-sticky custom event
+
+2014-03-25 11:44:27 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: only expose streams on dataflow
+	  Only probe on buffers, we don't want to expose the streams on events.
+
+2014-03-25 11:36:40 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: copy sticky events to ghostpad
+	  When we expose internal pads as ghostpads, first copy the sticky events
+	  so that we have the caps and segment etc.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724712
+
+2014-03-24 14:25:43 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: srtp handling
+
+2014-03-25 10:23:00 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: set SSRC on caps if known
+
+2014-03-24 16:58:25 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: put caps on udpsrc instead of using the signals
+	  Try to avoid using the request-pt-map to get caps but set them directly
+	  on the udpsrc element. That way, the caps get nicely transformed as they
+	  pass through the different elements in the rtpbin, including the AUX and
+	  decoder/encoder elements.
+
+2014-03-24 15:35:09 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: use profile to set rtcp caps
+	  Use the negotiated profile to set x-rtcp or x-srtcp caps
+
+2014-03-24 15:34:26 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: set udpsrc to READY
+	  READY is enough to allocate ports now
+
+2014-03-24 14:25:28 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: improve caps handling
+	  Protect caps with the lock.
+	  Don't push the caps event from the set_property function but mark the
+	  pad for reconfiguration so that it will renegotiate and push the new
+	  caps event in the streaming thread.
+
+2014-03-24 15:15:34 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: open/close socket in NULL<->READY state
+	  We should open the socket when going to NULL<->READY and not in the
+	  start/stop vemthod, which is called in READY<->PAUSED. This makes it
+	  possible to allocate a socket without going to PAUSED (and starting the
+	  negotiation).
+
+2014-03-24 14:35:01 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: free caps in ptmap array
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726696
+
+2014-03-20 11:12:51 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: handle NULL rtpmap and parse error better
+
+2014-03-16 23:46:22 -0400  Olivier Crête <tester@tester.ca>
+
+	* configure.ac:
+	  configure: Don't check for gudev if video4linux2 is not present
+
+2014-03-16 23:19:55 -0400  Olivier Crête <tester@tester.ca>
+
+	* configure.ac:
+	  configure: Don't fail if gudev is not present
+	  PKG_CHECK_MODULES has the bad habit of failing the build if it doesn't
+	  get what it wants, prevent that.
+
+2012-11-02 13:33:13 +0100  Olivier Crête <olivier.crete@collabora.com>
+
+	* configure.ac:
+	* sys/v4l2/Makefile.am:
+	* sys/v4l2/gstv4l2.c:
+	* sys/v4l2/gstv4l2devicemonitor.c:
+	* sys/v4l2/gstv4l2devicemonitor.h:
+	  v4l2: Implement GstDeviceMonitor subclass
+	  https://bugzilla.gnome.org/show_bug.cgi?id=678402
+
+2013-08-12 11:49:21 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* ext/pulse/Makefile.am:
+	* ext/pulse/plugin.c:
+	* ext/pulse/pulsedevicemonitor.c:
+	* ext/pulse/pulsedevicemonitor.h:
+	  pulse: Add device monitors
+	  https://bugzilla.gnome.org/show_bug.cgi?id=678402
+
+2014-03-16 19:24:26 -0400  Olivier Crête <tester@tester.ca>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Remove GstPropertyProbe leftovers
+
+2014-02-19 03:04:03 +0100  Mathieu Duponchelle <mduponchelle1@gmail.com>
+
+	* gst/videomixer/videomixer2.c:
+	* gst/videomixer/videomixer2.h:
+	  videomixer: Port to new collectpads API
+	  See: https://bugzilla.gnome.org/show_bug.cgi?id=724705
+
+2014-03-16 15:26:04 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/ext/types-compat.h:
+	* sys/v4l2/ext/videodev2.h:
+	  v4l2: Add types compatiblity for other OS
+	  Adds type compatiblity with other OS like BSD. This uses types mapping macro to
+	  avoid conflict with existing defined types. We resuse glib types as these are
+	  already available on supported platforms. This is GCC only because of the
+	  le32 type that uses bitwise attribute.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=726453
+
+2014-03-16 15:55:00 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/pulse/pulseutil.c:
+	  pulse: fix format info to caps conversion for mulaw
+
+2013-08-13 12:10:42 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* ext/pulse/pulsesink.c:
+	* ext/pulse/pulseutil.c:
+	* ext/pulse/pulseutil.h:
+	  pulse: Make gst_pulse_format_info_to_caps() shared
+	  https://bugzilla.gnome.org/show_bug.cgi?id=678402
+
+2014-03-15 18:41:16 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/Makefile.am:
+	  v4l2: Fix typo V4L_DIR intead of V4L2_DIR
+
+2014-03-15 18:05:32 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	  v4l2-build: Set HAVE_GST_V4L2 if headers are present
+	  The name of HAVE_ need to match the USE_. Now set HAVE_GST_V4L2 if
+	  videodev2.h is found.
+
+2014-03-15 16:47:51 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	* sys/Makefile.am:
+	  v4l2: Actually build the plugin
+	  The checks were removed inadvertedly in previous patch and not replaced.
+	  Re-introduce the configure checks and some of the checks in order to enable
+	  this plugin again. We only check if videodev2.h exist on the platform to
+	  avoid building on Windows or OSX, though we build against our own copy. This
+	  was breaking the build on built-bot.
+
+2014-03-15 13:47:42 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* 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:
+	  translation: PO file changes caused by POTFILE.in update
+
+2014-03-15 13:17:21 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	* po/POTFILES.in:
+	* po/POTFILES.skip:
+	* sys/v4l2/Makefile.am:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2videooverlay.c:
+	* sys/v4l2/gstv4l2videooverlay.h:
+	  v4l2: Remove XV support
+	  XV support for v4l2 never became upstream and ended up being
+	  commented out with an undef for a long time now.
+
+2014-03-15 11:13:05 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* configure.ac:
+	* gst-plugins-good.spec.in:
+	* sys/Makefile.am:
+	* sys/v4l2/ext/v4l2-common.h:
+	* sys/v4l2/ext/v4l2-controls.h:
+	* sys/v4l2/ext/videodev2.h:
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2vidorient.c:
+	* sys/v4l2/v4l2_calls.c:
+	* tests/icles/Makefile.am:
+	  v4l2: Use a copy of videodev2.h header
+	  With years the amount of ifdef have grown up and we are not even sure if the
+	  old code path compiles. Each time we need to update the v4l2 framework to add
+	  the new feature, we break compilation on older kernel. With exception of two
+	  controls in the video orientation control, this patch get rid of all ifdef by
+	  including the latest version of videodev2.h inside GStreamer.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723446
+
+2014-03-12 15:32:55 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: Add properties for selecting SSL/TLS certificate checking
+	  And by default properly check certificates against the system's CA
+	  certificates. Everything else is not a good default at all.
+
+2014-03-11 14:56:30 +0100  Per x Johansson <perxjoh@axis.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: fix assert on fps lower than 1
+	  Fixes assert caused by gst_duration_to_fraction calling
+	  gst_util_uint64_scale_int with a denominator of 0 when fps is less
+	  than 1.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=726106
+
+2014-03-11 00:46:06 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer2: store video info with buffers to keep it in sync
+	  Instead the queued buffer might have an old caps while the pad
+	  is already storing the information for a new caps. Mixing those
+	  while handling buffers will often lead to issues
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725948
+
+2014-03-08 19:29:58 -0500  William Manley <will@williammanley.net>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Fix typo contol -> control
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725632
+
+2014-03-04 01:15:49 +0000  William Manley <will@williammanley.net>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Normalise control names in the same way as v4l2-ctl
+	  V4L2 kernel drivers allow configuration of the hardware settings via a
+	  mechanism called controls.  These can be referred to by name such as
+	  "Brightness" and "White Balance Temperature".  The user-space command line
+	  client for setting these controls (v4l2-ctl) normalises these names such
+	  that they only contain lower case alphanumeric characters and the
+	  underscore '_'.  e.g:
+	  Kernel                     v4l2-ctl
+	  ----------------------------------------------------
+	  Brightness                 brightness
+	  White Balance Temperature  white_balance_temperature
+	  Focus (absolute)           focus_absolute
+	  GStreamer seems to want to follow this pattern but failed for controls with
+	  more than one consecutive non-alphanum character.  e.g. GStreamer would
+	  produce "focus__absolute_" rather than "focus_absolute".
+	  This commit fixes that issue.  Backwards compatibility is preserved by
+	  normalising all control names before comparison.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725632
+
+2014-03-07 16:17:29 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Make sure to not return EOS immediately if we finished a range request
+	  Only return EOS the next time create() is called, if at all. basesrc
+	  should already take care of not calling it again.
+	  Also always return immediately if the previous flow return was
+	  not OK. This indicates an error somewhere.
+
+2014-03-06 12:06:43 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpamrdepay.c:
+	* gst/rtp/gstrtpilbcdepay.c:
+	* gst/rtp/gstrtpsirendepay.c:
+	* gst/rtp/gstrtpspeexdepay.c:
+	  rtp: Remove caps restrictions from RTP depayloader sink caps
+	  Remove caps restrictions that correspond to the default and are not
+	  required in SDP. With the new usage of having pads require a subset
+	  of the caps, they will make the negotiation fail.
+
+2014-03-06 11:02:09 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtp/gstrtpspeexdepay.c:
+	  rtpspeexdepay: Remove caps restrictions for depayloader
+	  The "encoding-params" is optional in the SDP, because we now require
+	  a subset of the caps, it would fail caps negotiatioin if it wasn't present.
+	  So removed it from the template caps.
+
+2014-03-06 13:38:09 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Don't forget to quit mainloop after we cancelled when we got data after the stop position
+
+2014-03-06 13:35:47 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: If we had a stop position, allow for the server to finish our connection instead of just cancelling
+	  Otherwise keep-alive does not make much sense and also the server will have
+	  confusing things in the logs.
+
+2014-03-06 12:24:01 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: skip streams with same control url
+	  Keep track of what streams we did the SETUP for. We only need to
+	  configure caps, wait for pads and push events on setup streams. We can
+	  remove the disabled state of the stream and simplify some checks.
+	  After we setup a stream, skip the other streams that have the same
+	  control url. Use a skipped flag to mark streams that should be skipped.
+
+2014-03-06 12:22:47 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: remove obsolete code
+
+2014-03-05 16:19:19 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: just use the SDP index as the stream id
+	  Use the index of the media stream in the SDP as the stream id instead of
+	  keeping a separate counter.
+
+2014-03-05 13:35:19 +0100  Thijs Vermeir <thijsvermeir@gmail.com>
+
+	* sys/osxvideo/cocoawindow.m:
+	* sys/osxvideo/osxvideosink.m:
+	  osxvideo: fix build on Mac OSX Mavericks and put new window in front
+	  GetCurrentProcess/SetFrontProcess/TransformProcessType was deprecated
+	  and now removed in Mac OSX 10.9. orderFrontRegardless is used to make
+	  the video window the most front window.
+
+2014-03-05 17:33:56 +0100  Christian Fredrik Kalager Schaller <uraeus@linuxrising.org>
+
+	* gst-plugins-good.spec.in:
+	  Add docs directory to spec file
+
+2014-03-05 15:44:25 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: handle NULL control urls better
+
+2014-03-05 14:28:26 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  session: small cleanups
+	  It's nicer to explicitly check for NULL on pointer types to make it
+	  clear that it's a pointer and not a boolean.
+
+2014-03-05 14:26:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  session: handle unknown SSRC in FIR
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725712
+
+2014-03-05 11:39:09 +0100  Alessandro Decina <alessandro.d@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: fix seeking
+	  Call gst_rtspsrc_connection_flush (src, FALSE) to reset connections as
+	  non-flushing before sending PAUSE and PLAY with the new npt range. Without this
+	  patch, those commands would fail with EINTR as the connections were still
+	  flushing.
+
+2014-03-03 16:39:26 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/avi/gstavidemux.c:
+	* gst/avi/gstavidemux.h:
+	  avidemux: expose xsub as a subtitle instead of as a video
+	  It is placed inside a 'vids' struct, so it was being exposed on
+	  a pad named video_%d. XSUB are subtitles and this patch adds
+	  an special case for it to be exposed in a subpicture_%d pad
+
+2014-03-03 16:38:45 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: do not try to add a tag with tag_name set to NULL
+	  This can happen if there are subtitles in the stream, leading to
+	  an assertion
+
+2014-03-04 16:40:34 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: Add support for multiple payload types
+	  A media stream can have multiple payload types. Parse all the payload
+	  types and collect the caps information. We then have to store the
+	  pt<->caps mapping instead of 1 pt and 1 caps.
+	  Parse the profile from the SDP and use that to negotiate the transport
+	  instead of always using AVP.
+	  Rework how we do some tweaks for ASF and Realmedia.
+
+2014-03-04 11:34:39 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: refactor payload handling
+
+2014-03-03 11:34:00 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  jitterbuffer: fix buffer level with invalid DTS
+	  It is possible that the DTS is invalid (when we receive RTP packets from
+	  TCP, for example). As a fallback, use the reconstructed PTS value to
+	  calculate the buffer level.
+
+2014-03-02 05:10:13 +0100  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* .gitignore:
+	  .gitignore: Ignore gcov intermediate files
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725480
+
+2014-02-28 09:34:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From fe1672e to bcb1518
+
+2014-02-27 23:15:04 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  Revert "aacparse: put codec data on caps for loas format"
+	  This reverts commit e459cf3e01a08f1a3ef1fb954a41cfa36b3e510c.
+	  This was pushed by accident, the bug should likely be fixed in
+	  libav https://bugzilla.libav.org/show_bug.cgi?id=644
+
+2014-02-27 18:55:04 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: mark all parsed frames as sync points
+	  all jpeg frames are sync points, so mark them as such so
+	  reverse playback can properly work with the video decoder
+	  base class
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725104
+
+2014-02-25 01:12:05 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: put codec data on caps for loas format
+	  gst-libav audio decoder also needs codec data for LOAS format, otherwise
+	  it will complain about not having a decoder config and skip all packets
+	  https://bugzilla.gnome.org/show_bug.cgi?id=596772
+
+2014-02-27 00:43:48 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: align raw audio memory to powers of two
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725008
+
+2014-02-27 00:37:20 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: calculate alignment properly for audio depths not a multiple of 8
+
+2014-02-23 19:09:24 +0100  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: fix crash with 24-bit raw audio
+	  Do not try to align audio buffers to odd numbers,
+	  which will get us a NULL buffer which we then
+	  crash on.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725008
+
+2014-02-27 00:11:42 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtpmanager/Makefile.am:
+	  rtpmanager: re-enable -Werror
+
+2014-02-27 00:11:11 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: fix compiler warning
+	  gstrtpjitterbuffer.c: In function 'gst_rtp_jitter_buffer_loop':
+	  gstrtpjitterbuffer.c:2978:3: error: 'result' may be used uninitialized in this function
+	  while (result == GST_FLOW_OK);
+	  ^
+
+2014-02-26 22:11:41 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From 1a07da9 to fe1672e
+
+2014-02-26 21:11:23 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix uninitialized variable compiler warning
+
+2014-02-26 07:32:32 -0500  Jake Foytik <jake.foytik@ipconfigure.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Remove raw comparisons of RTP sequence numbers
+	  Several conditional statements perform comparison on RTP sequence
+	  numbers without taking the sequence number rollover into account.
+	  Instead, use the gst_rtp_buffer_compare_seqnum function to perform the
+	  comparison.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725159
+
+2014-02-03 01:44:21 +0100  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* tests/check/Makefile.am:
+	  tests: Don't build disabled plugins' check tests
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723502
+
+2014-02-26 11:29:45 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* docs/Makefile.am:
+	  docs: install prebuilt plugin docs if gtk-doc is disabled
+	  Sync to the Makefile.am from gst-plugin-base where it is done right.
+	  Fixes #725034
+
+2014-02-25 16:10:54 -0500  Hugues Fruchet <hugues.fruchet@st.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: do not emit "parsed" caps for vp8
+	  VP8 doesn't require parsing (vp8parse doesn't exist, so negotiation with demux fails
+	  if "parsed" is set in caps).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724636
+
+2014-02-11 16:27:08 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Don't require parser for VP8
+	  Until GStreamer has one (see bug722760), we should not require a parser for VP8.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722128
+
+2014-02-10 17:08:25 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: CAPTURE_MPLANE is well tested now
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722128
+
+2013-12-18 09:56:35 +0100  Benjamin Gaignard <benjamin.gaignard@linaro.org>
+
+	* sys/v4l2/gstv4l2.c:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2videodec.c:
+	* sys/v4l2/gstv4l2videodec.h:
+	  v4l2videodec: Create one element per device
+	  For each videoCdevice probe it input/output capabilities
+	  if it match with video decoder requirement register a new element.
+	  Signed-off-by: Benjamin Gaignard <benjamin.gaignard@linaro.org>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722128
+
+2013-12-19 15:26:52 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: Calculate latency from device information
+	  Decoders or other devices that expose a minimum buffers required produce
+	  an first output. We use this information to calculate latency.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722128
+
+2013-11-28 17:14:18 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/Makefile.am:
+	* sys/v4l2/gstv4l2.c:
+	* sys/v4l2/gstv4l2videodec.c:
+	* sys/v4l2/gstv4l2videodec.h:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2videodec: Implement v4l2videodec
+	  Implement an element that can driver V4L2 M2M decoder device.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722128
+
+2014-02-11 12:41:29 +0100  Göran Jönsson <goranjn@axis.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	  rtph264pay: only update last_spspps time if all sps/pps got sent successfully
+	  This fixes an issue with gst-rtsp-server where no sps and pps are
+	  sent for the first intra frame, because the payloader starts working
+	  already when receiving DESCRIBE but there is no transports so it tries
+	  to send sps and pps, but that fails with a FLUSHING flow. But the time
+	  for last sent sps and pps would still be set, so when PLAY arrives and
+	  the first intra frame is to be sent there is no sps and pps sent due to
+	  that time since last sps pps is less than spspps_interval.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724213
+
+2014-02-25 09:00:45 +0100  Santiago Carot-Nemesio <sancane@gmail.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix deadlock when task creation is no successful
+	  https://bugzilla.gnome.org/show_bug.cgi?id=725124
+
+2014-02-22 20:19:49 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/autodetect/gstautodetect.c:
+	  autodetect: demote candidate error to warning and plug fake{sink,src}
+	  In the case where we have no suitable candidate we post a warning and plug a
+	  fake-element. Do the same when non of the candidate work.
+	  This is more consistent and plugin the fakesink as a fallback is probably
+	  helpful for running unit tests without requiring hardware src/sink elements.
+	  Fixes #722981
+
+2014-02-23 12:34:48 +0100  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: make some more controls configurable
+	  ... at least if one tries hard enough using extra-controls property.
+
+2014-02-23 10:39:20 +0100  Dan Kegel <dank@kegel.com>
+
+	* configure.ac:
+	  v4l2: Require mplanar support for now in configure
+	  The code fails to compile without currently, see
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723446
+	  It's better to disable it instead of failing compilation
+	  until this is fixed properly.
+
+2014-02-23 00:14:04 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* ext/jack/gstjackaudioclient.c:
+	  jack: add some simple log handlers for jack
+	  Add log handlers for jack that write to the gst debug log. This avoids spamming
+	  the console when e.g. using autoaudiosink, having the jack elements installed,
+	  but not running jack.
+
+2014-02-22 21:31:21 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2src: handle old and odd driver behaviour when listing controls
+
+2013-11-28 16:54:58 -0800  Darryl Gamroth <dgamroth@uvic.ca>
+
+	* gst/audiofx/audiofxbaseiirfilter.c:
+	  audiofxbaseiirfilter: check if coefficients are provided inside filter lock
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719524
+
+2014-02-21 19:46:44 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2src: also unset INTERLACED flag on buffers if frame is not interlaced
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724899
+
+2014-02-21 14:31:59 +0000  Simon Farnsworth <simon.farnsworth@onelan.co.uk>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2src: Flag interlaced buffers as interlaced.
+	  We correctly indicate the field ordering on interlaced buffers, but fail to
+	  flag them as containing interlaced video, which we need to do here because
+	  we signal interlace-mode=mixed in our caps. This means that downstream
+	  elements (like vaapipostproc from gstreamer-vaapi) don't recognise these
+	  buffers as in need of deinterlacing.
+	  Fix this by setting the interlaced flag on all interlaced buffers.
+	  Signed-off-by: Simon Farnsworth <simon.farnsworth@onelan.co.uk>
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724899
+
+2014-02-19 13:56:37 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: be more strict at ADTS header parsing
+	  Adds two extra checks:
+	  - Sampling frequency on header can't be 15.
+	  - Frame size should be at least 9 or 7, depending
+	  on whether CRC protection is present.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724638
+
+2014-02-19 13:35:59 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: make sure we have enough ADTS data
+	  We need at least 6 bytes to pass over to _get_frame_len()
+	  but we were just checking for a minimum of 2 bytes for the
+	  syncword.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724638
+
+2014-02-20 22:52:57 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/autodetect/gstautodetect.c:
+	* gst/autodetect/gstautodetect.h:
+	  autodetect: check if the kid has a sync property
+	  previously autovideosrc did not have a sync property and v4l2src has none either.
+
+2014-02-19 21:55:52 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/autodetect/gstautoaudiosink.c:
+	* gst/autodetect/gstautoaudiosink.h:
+	* gst/autodetect/gstautoaudiosrc.c:
+	* gst/autodetect/gstautoaudiosrc.h:
+	* gst/autodetect/gstautodetect.c:
+	* gst/autodetect/gstautodetect.h:
+	* gst/autodetect/gstautovideosink.c:
+	* gst/autodetect/gstautovideosink.h:
+	* gst/autodetect/gstautovideosrc.c:
+	* gst/autodetect/gstautovideosrc.h:
+	  autodetect: use a common baseclass
+	  This makes the actual elements super simple. We're using the ELEMENT_FLAG to
+	  configure source/sink and a string for the Audio/Video type.
+
+2014-02-14 17:14:42 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: add tls-database property
+	  Add support for a new property: tls-database. If the property is set,
+	  the certificate database will be given to the rtsp connection if TLS
+	  protocol is being used. If the server certificate can't be verified with
+	  the default database, this additional database will be used.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724396
+
+2014-02-19 22:21:54 +0100  Thijs Vermeir <thijsvermeir@gmail.com>
+
+	* sys/osxaudio/gstosxaudioringbuffer.c:
+	* sys/osxaudio/gstosxaudiosink.c:
+	  osxaudio: remove unused variables
+
+2014-02-19 21:26:03 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/autodetect/gstautoaudiosink.c:
+	* gst/autodetect/gstautoaudiosrc.c:
+	* gst/autodetect/gstautodetect.c:
+	* gst/autodetect/gstautodetect.h:
+	* gst/autodetect/gstautovideosink.c:
+	* gst/autodetect/gstautovideosrc.c:
+	  autodetect: extract common helper code
+	  The function to generate the pretty names is basically the same. Use one and add
+	  a parameter.
+
+2014-02-19 21:01:39 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/autodetect.c:
+	  autodetect: improve the tests
+	  Add fake audio/video sinks. Previously running the test might be flaky due to
+	  the use of real elements (hardware in use), which we don't want to test here.
+	  Add two more tests that check that the fakes are chosen.
+
+2014-02-19 15:19:30 +0100  Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: do not emit error when connection with unknown size ends
+	  Commit 46fd12ae5ec53200b16dfd7f17048d6bc60fbfbc introduced connection
+	  recovery. But when server does not specify content-size,
+	  souphttpsrc tries to reconnect even after regular end of stream.
+	  Http server replies  with SOUP_STATUS_REQUESTED_RANGE_NOT_SATISFIABLE
+	  but souphttpsrc still emits error instead of EOS.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=724717
+	  Signed-off-by: Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+2014-02-19 11:26:22 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* tests/check/elements/autodetect.c:
+	  autodetect: fix the disabled test
+	  Use a shared helper for both tests. It turns out that the valgrind variant is
+	  fine (maybe due to picking up pulsesink though).
+
+2014-02-19 11:05:35 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* tests/check/elements/autodetect.c:
+	  autodetect: remove cruft from the test
+	  Remove the obsolete version check and use the ignore macro for the disabled test.
+
+2014-02-18 22:54:45 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/audiofx/audiofirfilter.c:
+	* gst/audiofx/audioiirfilter.c:
+	* gst/level/gstlevel.c:
+	* gst/spectrum/gstspectrum.c:
+	  docs: use docbook markup for xi:include
+	  It turns out that the change in gtk-doc-1.20 which wraps the |[]| content in
+	  CDATA break xi:inlcude examples. As in a whole jhbuild checkout these where
+	  the only 4, we're fixing them instead.
+
+2014-02-18 22:35:45 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/isomp4/gstqtmux-doc.h:
+	  isomp4mux: fix copy and paste
+	  This fixes doc warnings.
+
+2014-02-18 21:44:24 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/debugutils/gstcapssetter.c:
+	* gst/isomp4/gstqtmux-doc.c:
+	* gst/isomp4/gstqtmux.c:
+	* gst/level/gstlevel.c:
+	* gst/replaygain/gstrganalysis.c:
+	* gst/replaygain/gstrgvolume.c:
+	  docs: use the gtk-doc syntax to link to properties
+	  Don't use docbook unless needed. Also stip other docbook tags in the the files we fix.
+
+2014-02-18 11:28:18 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: fix crash when getting the current-device in NULL->READY
+	  The "goto unlock" is wrong as in this code path we haven't take the lock yet.
+	  Fixes #724619
+
+2014-02-14 22:50:49 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  soup: We need libsoup >= 2.40 for proper usage of the content decoder
+	  Previous versions did not consider our chunk allocator and allocated
+	  memory by themselves, which caused crashes and broken behaviour.
+
+2014-02-14 15:27:20 -0500  William Jon McCann <william.jon.mccann@gmail.com>
+
+	* gst/audiofx/audiocheblimit.c:
+	* gst/udp/gstudpsrc.c:
+	  docs: fix mismatched para tags
+	  newer gtkdoc is more sensitive to mismatched docbook tags.
+	  This fixes the build in master.
+
+2014-02-14 15:59:46 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: add support for serialized queries
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=723850
+
+2014-02-14 15:53:55 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: fix typecast to fix compilation
+
+2014-02-14 12:01:00 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpsession: proxy caps and allocation on RTP pads
+	  recv_rtp_sink: allow proxying of the allocation query.
+	  send_rtp_sink: allow proxying of caps and allocation. This allows us to
+	  query caps downstream as well as get an allocator from downstream.
+	  send_rtp_src: allow proxy of caps, this makes the caps query do
+	  upstream.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=723850
+
+2014-02-13 12:29:13 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: handle tags in mac encoding
+	  Check the charset from (C)*** tags and set the charset
+	  to convert from MAC encoding if suitable.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723166
+
+2014-02-13 12:09:13 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Use new automatic_eos API from basesrc
+	  We want to notice ourselves that we're EOS. Otherwise we will
+	  always cancel requests in the very end and confuse the server...
+	  and also make it impossible to use persistent connections.
+
+2014-02-13 11:11:13 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Consistently use have_size instead of content_size!=0
+
+2014-02-13 10:30:09 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Free extra headers when finalizing the element
+	  It's set as property by the application, we should not just reset
+	  properties when going back to READY.
+
+2014-02-13 10:28:13 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Properly close the session when going back to NULL
+	  Don't wait for that until the element is disposed.
+
+2013-02-28 12:20:52 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: add support for keep-alive sessions
+	  https://bugzilla.gnome.org/show_bug.cgi?id=699926
+
+2014-02-12 13:00:13 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: Add "compress" property to enable/disable automatic gzip/deflate content encoding handling
+
+2014-02-12 12:39:10 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Retry connection if we're finished before the content size only if we actually have a content size
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722185
+
+2014-02-12 10:08:50 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouputils.c:
+	  souputils: Fix compiler warning
+	  gstsouputils.c:35:25: error: comparison of constant 9 with expression of type
+	  'SoupLoggerLogLevel' is always false
+	  [-Werror,-Wtautological-constant-out-of-range-compare]
+
+2014-01-07 23:00:56 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* ext/soup/Makefile.am:
+	* ext/soup/gstsoup.c:
+	* ext/soup/gstsouphttpclientsink.c:
+	* ext/soup/gstsouphttpclientsink.h:
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	* ext/soup/gstsouputils.c:
+	* ext/soup/gstsouputils.h:
+	  souphttp*: add ability to do HTTP session logging
+	  This changeset adds the loggin infrastructure and
+	  mods both souphttpsrc and souphttclientsink to use it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721764
+
+2014-02-07 14:00:15 +0100  divhaere <dirk.vanhaerenborgh@ugent.be>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-mux.c:
+	  matroska: add support for GRAY8, BGR and RGB video colourspaces in V_UNCOMPRESSED codec
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723849
+
+2014-02-11 13:25:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Add mapping for NOT_FOUND and NOT_AUTHORIZED errors
+
+2014-02-11 13:25:22 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Don't duplicate status_code to GStreamer error mapping
+
+2014-02-09 23:38:44 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/goom/filters.c:
+	* gst/goom2k1/filters.c:
+	  goom: Remove unused functions
+
+2014-02-09 23:21:20 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-parse.c:
+	  matroskaparse: Comment out some unused functions used only from the commented out pull-mode code
+
+2014-02-08 21:01:32 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/taglib/gstid3v2mux.cc:
+	  id3v2mux: Fix another compiler warning
+
+2014-02-08 17:43:32 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  souphttpsrc: Fix implicit enum conversion compiler warning
+	  error: implicit conversion from enumeration type
+	  'SoupStatus' to different enumeration type 'SoupKnownStatusCode'
+
+2014-02-08 17:41:21 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/interleave.c:
+	  interleave: Fix unitialized variable compiler warning in test
+	  error: variable 'mask' is used uninitialized
+	  whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]
+
+2014-02-08 17:27:51 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/taglib/gstid3v2mux.cc:
+	  id3v2mux: Fix unitialized variable compiler warning
+	  error: variable 'image_type' is used uninitialized
+	  whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]
+
+2014-02-08 17:25:27 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* sys/oss4/oss4-audio.h:
+	  oss4: Fix typo in header include guard
+	  error: 'GST_OSS4_AUDIO_H' is used as a header guard here,
+	  followed by #define of a different macro [-Werror,-Wheader-guard]
+
+2014-02-08 17:24:06 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: Fix unitialized variable compiler warning
+	  variable 'rtx_ssrc' is used uninitialized whenever
+	  'if' condition is false [-Werror,-Wsometimes-uninitialized]
+
+2014-02-08 17:21:19 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpac3depay.c:
+	  rtpac3depay: Remove unused variable
+
+2014-02-08 17:19:19 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/flx/flx_fmt.h:
+	  flx: Fix typo in header include guard
+	  error: '__GST_FLX_FMT__H__' is used as a header guard here,
+	  followed by #define of a different macro [-Werror,-Wheader-guard]
+
+2014-02-07 10:07:41 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: remove have_dts flag from pads
+	  It was used in the past in 0.10 when there was no explicit DTS
+	  field in buffers, now we have it in 1.x series and we can
+	  check it directly with GST_BUFFER_DTS_IS_VALID
+
+2014-02-07 01:49:26 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: improve support for sparse streams
+	  Do not try to use subsequent buffer timestamps to calculate
+	  sparse streams durations because the stream is sparse and
+	  the buffers might not be 'time adjacent'. So rely on the
+	  duration and give the option to the pad to provide
+	  custom 'empty' buffers to represent the gaps in the
+	  stream, this can vary on how the data is represented.
+	  Right now, the only sparse stream supported is tx3g subtitles.
+
+2014-02-06 12:15:22 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	  qtmux: add support for text/x-raw subtitles
+	  Adds it to mp4mux, qtmux and gppmux.
+	  Buffers need to be prefixed with 2 bytes for the text length before
+	  being muxed.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=581295
+
+2014-02-06 12:09:01 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	  qtmux: add support for the TX3G atoms
+	  Adds functions for creating and setting values related to the
+	  tx3g atom for raw text subtitle support.
+	  QTFF spec has information on those atoms
+	  https://bugzilla.gnome.org/show_bug.cgi?id=581295
+
+2014-02-05 10:27:54 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	* gst/isomp4/gstqtmuxmap.c:
+	* gst/isomp4/gstqtmuxmap.h:
+	  qtmux: add subtitle support to qtmuxmap structures
+	  adds basic stubs for subtitle support around the qtmux and
+	  qtmuxmap structures. Still no real subtitle implemented, but
+	  basic functions in place
+	  https://bugzilla.gnome.org/show_bug.cgi?id=581295
+
+2014-01-20 17:31:14 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.c:
+	* gst/matroska/matroska-ids.h:
+	* gst/matroska/matroska-parse.c:
+	* gst/matroska/matroska-read-common.c:
+	* gst/matroska/matroska-read-common.h:
+	  matroska: factor out read context init/reset
+	  While at this, move _track_reset() to track-ids
+	  so it can be called from the common read context
+	  reset routine.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722705
+
+2014-02-06 12:21:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/effectv/gstrev.c:
+	  effectv: fix doc section of revtv element
+
+2014-02-05 12:46:54 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* sys/osxvideo/Makefile.am:
+	  osxvideo: Fix libtool usage
+	  --tag=CC is needed for static build
+
+2014-01-16 11:26:41 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: do not try set deinterlace method if passthrough is enabled
+	  Fixes an issue with progressive content and unsupported video formats
+	  for the deinterlace method.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719636
+
+2014-02-04 21:26:56 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: order format in template caps by preference
+	  To minimise risk of bad fixation, though audioconvert
+	  at least should be smart enough to avoid it.
+
+2014-02-02 09:57:03 -0800  Dan Kegel <dank@kegel.com>
+
+	* configure.ac:
+	  v4l2: Remove obsolete definition GST_V4L2_MISSING_BUFDECL
+	  The only use was removed by 9edc0c0365f79ab07ff2e65461c6696e3931a3f0
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723446
+
+2014-02-04 13:43:56 +0100  Rafał Mużyło <galtgendo@o2.pl>
+
+	* ext/flac/gstflacdec.c:
+	* ext/flac/gstflacenc.c:
+	* gst/cutter/gstcutter.c:
+	  gst: Don't use endianness-specific S8 audio format
+	  It does not exist.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723331
+
+2014-01-31 14:17:54 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* ext/cairo/gstcairooverlay.c:
+	  cairooverlay: add support for RGB16
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723289
+
+2014-01-30 09:43:50 +0100  Per x Johansson <perxjoh@axis.com>
+
+	* gst/matroska/matroska-mux.c:
+	* gst/matroska/matroska-mux.h:
+	  matroskamux: Fix constantly growing used uid list
+	  Moves the used uid list to the class to avoid having it grow forever.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723269
+
+2014-01-30 10:44:05 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From d48bed3 to 1a07da9
+
+2014-01-24 01:52:08 +0000  Mike Sheldon <elleo@gnu.org>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: Ignore Broadcast Wave Format (BWF) tags when searching for 'fmt' chunk
+	  https://bugzilla.gnome.org/show_bug.cgi?id=723125
+
+2014-01-29 10:37:53 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* tests/check/elements/rtpaux.c:
+	  check: Use fakesink sync=True instead of an audio sink
+	  Ensures the test can run on systems without alsa (or any audio output for
+	  that matter), and will avoid people running build slaves wondering what
+	  the hell was beeping during the night :)
+
+2014-01-27 20:05:42 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/audioparsers/gstac3parse.c:
+	  ac3parse: custom get_sink_caps handling for private stream caps
+	  ... now that those are transformed rather than parsed, some transforming
+	  of caps is required as well to make auto-plugging succeed.
+
+2014-01-25 02:06:00 -0500  Ryan Lortie <desrt@desrt.ca>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: guard use of ENODATA with #ifdef
+	  Not all systems with v4l have ENODATA defined, so check that we have it
+	  before attempting to use it.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722953
+
+2014-01-24 12:37:39 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  Revert "rtspsrc: Proxy rtpjitterbuffer do-retransmission property"
+	  This reverts commit 9f7b1128b1f00a2b87a232ff890867549ab95ba5.
+	  This should be handled automatically be rtspsrc if the AVPF profile
+	  is used, and manual enabling of it can be done with the new-manager
+	  signal.
+
+2014-01-24 10:21:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: add signal to notify of new manager
+	  So that you can configure and connect to signals on the rtpbin.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=722866
+
+2014-01-23 15:17:58 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: Proxy rtpjitterbuffer do-retransmission property
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722866
+
+2014-01-21 17:52:44 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: handle expected packet being an RTX packet
+	  If the expected packet (do_next_seqnum is TRUE) is the one we requested
+	  for retranmission earlier, do the logic to update the retransmission
+	  statistics as well before setting up the timers for the next expected
+	  packet.
+	  Also reset the retransmission counter if the timer is reused for another
+	  seqnum.
+
+2014-01-21 15:48:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: add a caps accumulator for the request-pt-map signal
+	  Add an accumulator that stops the signal emission as soon as a caps has
+	  been retrieved. Otherwise the default handler would continue emitting
+	  the signal and possibly overwrite the result with NULL again.
+
+2014-01-21 15:25:54 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtxreceive: copy flags and timestamps from original buffer
+
+2014-01-21 15:24:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: ignore invalid timestamps in rtt calculation
+	  When the input buffer does not have a valid timestamp, don't try to
+	  calculate the round-trip-time.
+
+2014-01-16 14:23:13 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-parse.c:
+	* gst/matroska/matroska-read-common.c:
+	* gst/matroska/matroska-read-common.h:
+	  matroskaparse: better default caps when none set
+	  Uses information gathered during EBML parsing to
+	  forge a more suitable set of caps instead of blindly
+	  assuming everything is video/x-matroska.
+	  For consistency, stream type reset was added to
+	  matroska-demux too.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722311
+
+2014-01-15 17:29:35 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests: rtprtx::test_rtxreceive_data_reconstruction: remove useless code for triggering retransmission
+	  There is no need anymore to push yet another buffer in rtxsend
+	  in order to trigger the previously requested retransmissions
+	  to actually happen.
+
+2014-01-15 17:27:19 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests: rtprtx::test_rtxreceive_data_reconstruction: fix race condition
+	  Now with rtprtxsend pushing rtx buffers from a different thread,
+	  this is necessary to ensure that the result of the test is deterministic.
+	  This code makes use of GstCheck's global GMutex and GCond that are
+	  being used inside GstCheck's sink pad chain() function in order
+	  to synchronize with it.
+
+2014-01-15 17:17:57 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests: rtprtx::test_rtxsender_packet_retention: fix race condition
+	  Now with rtprtxsend pushing rtx buffers from a different thread,
+	  this is necessary to ensure that the result of the test is deterministic.
+	  This code makes use of GstCheck's global GMutex and GCond that are
+	  being used inside GstCheck's sink pad chain() function in order
+	  to synchronize with it.
+
+2014-01-15 11:26:33 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests: rtprtx::test_push_forward_seq: fix race condition
+	  Now with rtprtxsend pushing rtx buffers from a different thread,
+	  this is necessary to ensure that the result of the test is deterministic.
+	  This code makes use of GstCheck's global GMutex and GCond that are
+	  being used inside GstCheck's sink pad chain() function in order
+	  to synchronize with it.
+
+2014-01-15 09:47:03 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests: rtprtx::test_push_forward_seq: fix buffer refcounting
+
+2014-01-21 13:42:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: ensure that no rtx buffers are sent after EOS
+	  To do that, enqueue the EOS event to be sent from the srcpad task
+	  thread and flush the queue right afterwards, so that no more rtx
+	  buffers can be sent, even if there are more requests coming in.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722370
+
+2014-01-15 09:46:14 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	  rtprtxsend: run a new GstTask on the src pad
+	  The reason behind this is to minimize the retransmission delay.
+	  Previously, when a NACK was received, rtprtxsend would put a
+	  retransmission packet in a queue and it would send it from chain(),
+	  i.e. only after a new buffer would arrive.
+	  This unfortunately was causing big delays, in the order of 60-100 ms,
+	  which can be critical for the receiver side.
+	  By having a separate GstTask for pushing buffers out of rtxsend,
+	  we can push buffers out right after receiving the event, without
+	  waiting for chain() to get called.
+
+2014-01-03 17:47:55 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/shout2/gstshout2.c:
+	* ext/shout2/gstshout2.h:
+	  shout2send: error out if no caps were received
+	  Instead of assuming that input is ogg.
+
+2014-01-03 17:30:12 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2send: accept audio/webm, audio/ogg and video/ogg as well
+	  Those are advertised in the template caps, but the
+	  setcaps handler didn't handle them. But then oggmux
+	  and oggparse at least for now still always output
+	  application/ogg anyway, so that wasn't a real problem.
+
+2014-01-20 10:12:45 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvp8pay.c:
+	  rtpvp8pay: Don't leak input buffers
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722414
+
+2014-01-19 17:40:56 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/avi/gstavimux.c:
+	  avimux: reset some more audio pad data when needed
+
+2014-01-19 17:38:59 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/avi/gstavimux.c:
+	* gst/avi/gstavimux.h:
+	  avimux: write correct blockalign for vbr audio
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720659
+
+2014-01-16 17:36:12 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: do not drop serialized events when latency is set
+	  Serialized events are now queued in the jitter buffer, so we don't
+	  want to drop them even latency is set.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722372
+
+2013-12-11 09:36:22 +0100  Michael Olbrich <m.olbrich@pengutronix.de>
+
+	* gst/avi/gstavimux.c:
+	  avimux: don't make the buffer writable unless absolutely necessary
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722396
+
+2013-09-12 16:56:56 +0200  Michael Olbrich <m.olbrich@pengutronix.de>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2: set GST_BUFFER_FLAG_DELTA_UNIT when appropriate
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722394
+
+2014-01-17 07:46:09 +0100  Michael Olbrich <m.olbrich@pengutronix.de>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: don't ref the newly created allocator
+	  Otherwise the allocator will never be deleted.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712612
+
+2014-01-15 22:47:12 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Don't skip all video frames until the first keyframe
+	  Instead do it like all other demuxers and let parsers and decoders
+	  handle that. The keyframe information inside the container might
+	  be completely wrong like in the sample file of the bug report,
+	  and if it is correct and we push no keyframes, then the parsers
+	  and decoders will handle that properly anyway.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=682276
+
+2014-01-13 10:08:09 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: remove elst_offset variables
+	  They are not used anymore
+
+2014-01-06 21:36:17 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: remember reverse playback when verifying the segment end
+	  Check if the rate is positive or negative to correctly compare the current
+	  position with the segment to make reverse playback work
+
+2014-01-03 10:59:35 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: do not ignore empty segments
+	  Make sure empty segments are used and pushed with a gap event
+	  to represent its data (or lack of it)
+	  Each QtSegment is mapped into a GstSegment with the corresponding
+	  media range. For empty QtSegments a gap event is pushed instead
+	  of GstBuffers and it advances to the next QtSegment.
+	  To make this work with seeks, need to keep track of the starting
+	  'base' to make sure it remains consistently increasing when
+	  pushing new segment events.
+	  For example: if a seek makes qtdemux start from 5s, the first
+	  segment will have a base=0. When the next segment is activated,
+	  its base time will be QtSegment.time - qtdemux.segment_base so
+	  that it doesn't include the first 5s that weren't played and
+	  shouldn't be accounted on the running time
+	  This purposedly will remove the fix made for
+	  https://bugzilla.gnome.org/show_bug.cgi?id=700264, at this
+	  point it was decided to respect the gaps, even if they cause
+	  a delay on playback, because that's the way the file was crafted.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=345830
+
+2013-12-12 23:05:43 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests: Remove usage of the system clock from the rtprtx test
+
+2013-12-12 23:22:41 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* tests/check/elements/rtpcollision.c:
+	  tests: Initial segment in rtpcollision test
+
+2014-01-14 15:56:42 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/examples/rtp/client-rtpaux.c:
+	* tests/examples/rtp/server-rtpaux.c:
+	  examples/*-rtpaux: specify payload type association for the audio stream, so that rtx works also for audio
+
+2014-01-14 13:08:18 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: remove wrong check for payload type not having been set
+	  1) pt can be lower than 96
+	  2) there is no point in checking that because rtprtxsend will not
+	  even store buffers for payload types that it doesn't know about,
+	  so this case will never be reached
+
+2014-01-14 13:01:41 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: fix data locking when creating rtx packets
+	  This patch moves the creation of rtx packets to be done early,
+	  in the src_event() function, when they are requested. The purpose
+	  is to run gst_rtp_rtx_buffer_new() with the object locked to
+	  protect internal data, because if it is done at the pushing stage,
+	  we would have to lock and unlock multiple times in a row while we
+	  are pushing the rtx buffers.
+	  Previously there was no locking at all, which was terribly wrong.
+
+2014-01-14 12:50:23 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: lock access to internal data in sink_event() function
+
+2014-01-14 12:44:06 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: remove unnecessary call to reset() from finalize()
+	  ...and use _free_full() on the pending buffers queue now that
+	  reset() is not being called
+
+2014-01-14 12:38:51 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: remove unused parameter from the internal reset() method
+
+2014-01-14 12:32:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: Use g_slice_* for allocating internal structures
+
+2014-01-14 12:28:01 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtprtxreceive: remove stupid mutex unlock in the middle of chain()
+
+2014-01-14 12:25:36 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtprtxreceive: use GST_DEBUG_OBJECT / GST_WARNING_OBJECT instead of GST_DEBUG / g_warning
+
+2014-01-14 12:19:58 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtprtxreceive: fix integer format specifiers in GST_DEBUG
+	  seqnum in this function is 32-bit, so G_GUINT16_FORMAT would
+	  produce undefined output on big endian systems
+
+2014-01-14 12:13:49 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	  rtprtxsend: change the rtx_pt_map directly in set_property() instead of delaying it for chain()
+	  The same lock is held, so there is no point in complicating it...
+
+2014-01-14 12:07:58 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/rtpmanager/gstrtprtxreceive.h:
+	  rtprtxreceive: change the rtx_pt_map directly in set_property() instead of delaying it for chain()
+	  The same lock is held, so there is no point in complicating it...
+
+2014-01-14 11:55:00 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	  rtprtxreceive: simplify the code of finalize()
+
+2014-01-14 11:52:21 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/rtpmanager/gstrtprtxreceive.h:
+	  rtprtxreceive: use the GstObject lock instead of a new one
+
+2014-01-14 11:45:52 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	  rtprtxsend: use the GstObject lock instead of a new one
+
+2013-12-10 14:29:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Add NV12_64Z32 support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722127
+
+2014-01-14 19:08:49 +0900  Justin Joy <justin.joy.9to5@gmail.com>
+
+	* sys/oss/gstosshelper.c:
+	  osshelper: Don't leak fd when getting card name
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722163
+
+2014-01-14 09:43:33 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/audioparsers/gstaacparse.c:
+	  Revert "aacparse: relax the detection of ADTS"
+	  This was pushed by mistake along with the V4L2 fix.
+	  This reverts commit 8eb4b032bef444397c4d211f2095c173ba114187.
+
+2014-01-14 15:42:01 +0900  Justin Joy <justin.joy.9to5@gmail.com>
+
+	* gst/rtp/gstrtpg726pay.c:
+	  rtpg726pay: don't leak encoding_name string
+	  https://bugzilla.gnome.org/show_bug.cgi?id=722159
+
+2014-01-13 09:14:00 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: fix build break using V4L2_CAP_VIDEO_M2M_MPLANE
+	  This may not be defined. Since the previous version used
+	  only the other define (V4L2_CAP_VIDEO_OUTPUT_MPLANE), fall
+	  back on this only when not available.
+
+2013-02-27 01:45:52 +0900  Akihiro Tsukada <atsukada@users.sourceforge.net>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: relax the detection of ADTS
+	  According to ISO/IEC 13818-7, "channel_config" field in ADTS header
+	  may have value of 0, as in the case of frame with PCE.
+	  gst_aac_parse_detect_streams() returned FALSE for those frames
+	  and discarded them.
+
+2014-01-07 11:58:23 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: check set_config return value in gst_v4l2_buffer_pool_new
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2014-01-10 12:40:31 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Add parsed=1 field for encoded output
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2014-01-10 12:39:16 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't leak empty caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2014-01-08 16:51:21 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: do not stop a stream not previously started
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-12 16:27:21 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't enforce dimension field on encoded formats
+	  Don't enforce having width, height and framerate in template caps for encoded
+	  formats. These don't always need to be exposed and may break negotiation for
+	  decoder and decoding sink. If needed, these field will be automatically added
+	  when probed caps are known.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-12 17:09:59 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: unref downstream pool
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-18 13:37:23 -0500  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	  v4l2bufferpool: add gst_v4l2_buffer_pool_flush
+	  STREAMOFF set all v4l2buffers to DEQUEUE state.
+	  Then for CAPTURE we call QBUF on each buffer.
+	  For OUTPUT the buffers are just push back in the GstBufferPool
+	  base class 's queue.
+	  But the loop actually looks like the same.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-16 17:29:30 -0500  Benjamin Gaignard <benjamin.gaignard@linaro.org>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Add vp8 support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-12 16:46:09 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't force framerate field for OUTPUT
+	  If there is nothing that seems to force a certain framerate on output device, it is
+	  preferable to simply not set that feild. This allow negotiation with tsdemux in a
+	  decoder for example.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-12 14:07:03 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: _v4l2fourcc_to_structure() can be static
+	  This function is not used anymore outside v4l2object.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-12 14:22:26 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Add MPEG1/2 support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-12 12:18:45 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Ask for a decent buffer size when dealing with encoded formats
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-07 14:03:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: On warn on size change if n_planes > 1
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-31 16:38:09 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: check if translated format is valid
+	  Also add a FIXME in gst_v4l2_object_setup_format
+	  to note that the whole function has to be improved
+	  in order to support ENCODED formats.
+	  It requires to have an encoder device which we do not
+	  have right now.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-07 10:31:15 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Validate returned dimensions
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-05 19:36:25 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Ensure max is not smaller then min in decide_allocation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-05 19:36:06 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't keep the max paramter when using our own pool
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-05 19:34:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: Respect the suggested min buffer
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-05 18:48:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Allocate pool if needed in decide_allocation
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-05 18:49:19 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Add V4L2_CID_MIN_BUFFERS_FOR_CAPTURE support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-05 18:48:15 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2: Move decide allocation into v4l2object
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-05 13:51:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Implement _setup_format()
+	  This method allow setting up the object from the currently configured format on the
+	  device. This is useful for M2M element where input data decides the format that will
+	  be set on capture side.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-10 14:34:17 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Split out saving format from set_format()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-31 15:37:26 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: set only one plane for encoded format
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-04 16:49:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Move code block where it belongs
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-04 16:26:12 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't check format specific information
+	  The number of plane, and the stride does not represent a capability change. Same caps
+	  can have different stride from the default GstVideoInfo and the number of planes will
+	  never change for 1 format.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-04 16:23:18 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Move the extrapolation of stride at the right place
+	  Now that we have a stride array, we should extrapolate only when
+	  eeded (non multi-planar buffer).
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-04 15:09:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Move back assertions where they should be
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-04 15:09:10 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Move mplane logic into gst_v4l2_object_get_caps_info()
+	  It makes the gst_v4l2_object_set_format() slightly simplier and will make that
+	  logic reusable. Note that gst_v4l2_object_has_mplane() will always return the
+	  same value for one device. There is no need to check against the caps as this
+	  has already been done by _open.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-03 18:27:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Split _v4l2fourcc_to_video_format
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-02 18:05:11 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: Request buffers only once
+	  VIDIOC_REQBUFS allocates buffer, it has no place inside set_config. Also, some driver do
+	  no allow multiple calls to this ioctl.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-02 15:26:50 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Don't validate dimension for encoded format
+	  We set the dimensions just in case but don't validate them
+	  afterwards. For some codecs the dimensions are *not* in the
+	  bitstream, IIRC VC1 in ASF mode for example.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-11-28 17:10:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Quirks for dev without initial format
+	  Most M2M have undefined behaviour initially when VIDIOC_G_FMT is called.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-11-28 17:09:26 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Add gst_v4l2_object_open_shared()
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-11-28 17:07:05 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/v4l2_calls.c:
+	* sys/v4l2/v4l2_calls.h:
+	  v4l2object: Implement gst_v4l2_dup()
+	  This will duplicated the FD from another object and copy over the probed result.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-11-28 16:59:59 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: make IO_MODE enum public
+	  This is to allow adding a second io-mode property on M2M device like decoder so
+	  input and output can be controlled separatly.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-06-04 23:42:24 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: better handle quirks activation
+	  This way we can activate deactivate those quirks all at once at one
+	  place.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-06-04 23:34:04 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Fix h264 caps
+	  V4L2_PIX_FMT_H264 is documentated as byte-stream (with start code). The ensure proper
+	  negotiation with element like h264parse.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2013-12-06 14:44:51 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Split caps in different categories
+	  This is need to correctly expose capabilities on specialized devices
+	  like decoders and encoders.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720568
+
+2014-01-10 14:16:00 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-read-common.c:
+	  matroskademux: don't leak TOC chapter list
+
+2014-01-10 08:52:16 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: remove obsolete write-dummy-and-overwrite-on-eos code
+	  The need for rewriting apparently is obsolete 0.10 leftover.
+	  We now have caps for subtitles when we create the headers,
+	  so we always write the correct data in the first place.
+
+2014-01-09 23:55:16 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: remove duplicate assignment
+	  Coverity CID 1151680
+
+2014-01-09 18:25:04 +0000  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: write subtitle codec ID and data at start when known
+	  This avoids issues with writing dummy data first, then having
+	  to come back and write correct data later. Doing so prevents
+	  the muxed stream from being actually streamable.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712134
+
+2014-01-09 17:32:15 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  configure: Include AvailabilityMacros.h for osxvideo check
+	  Otherwise MAC_OS_X_VERSION_MIN_REQUIRED might not be defined
+
+2014-01-09 11:56:31 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	  qtmux: respect the HDLR box string format for mov and isomedia
+	  Mov spec says it uses a pascal style string, while isomedia uses
+	  a null terminated one. Store the current atoms flavor into the HDLR
+	  to be able to generate the correct output.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=705982
+
+2014-01-08 11:28:04 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/matroska/matroska-mux.c:
+	  Revert "matroskamux: Use the running time for container timestamps, not buffer timestamps"
+	  This reverts commit b3aa8755fe07639f22e4104f4932d769d6c9075a.
+	  We are already using the running-time because they were placed on the
+	  buffers with gst_collect_pads_clip_running_time(). Arguably it would be
+	  better to not modify the incomming buffers but collectpads seems to want
+	  to use absolute timestamps from the buffers for finding the best buffer
+	  (this can be changed with a custom compare function..).
+
+2014-01-08 10:41:24 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  configure: Fix AC_COMPILE_IFELSE usage
+
+2014-01-08 10:31:18 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  osxvideosink: Improve configure check for OSX >= 10.6
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721245
+
+2014-01-07 12:13:51 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: remove unused list of decoders
+	  remove list of decoders, which are already handled by the list of elements.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719938
+
+2014-01-08 09:46:55 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Error out if ADPCM caps don't contain the layout field
+
+2014-01-03 15:25:23 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Add support for g726 ADPCM
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720995
+
+2014-01-07 15:04:02 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: use new method to get media-type
+	  Use the new method to get the media type of a transport.
+
+2014-01-06 21:12:17 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* tests/check/elements/wavparse.c:
+	  wavparse: split the test
+	  This way one failure won't shadow the other test and also if one fails we get
+	  better disgnostics through the test-name.
+
+2014-01-06 14:54:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Add HEVC / h265 support
+
+2014-01-06 14:54:38 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.h:
+	  matroskademux: Add HEVC / h265 support
+
+2014-01-06 13:36:38 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: remove ifdef'ed code
+	  We do have adtl and cue parse as part of toc handling alreday. The fmt code is a left over from <0.10 times.
+
+2014-01-06 13:32:58 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/avi/gstavidemux.c:
+	* gst/wavparse/gstwavparse.c:
+	  avidemux, waveparse: more logging for unhandled chunks
+	  Always print a warning with the tag and if possible do a memdump.
+
+2014-01-05 22:47:42 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: expose 'strn' - stream name - as title tag
+
+2014-01-05 22:41:24 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: parse fuji strd
+	  We can get maker, model and capture date from this chunk.
+	  Fixes #636143
+
+2014-01-05 21:46:33 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: ... and use the local api both times
+
+2014-01-05 21:38:14 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: copy the riff api for ncdt into the element
+	  This chunk is avi specific, no need to expose this as public api.
+
+2014-01-05 10:28:21 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Add missing semicolon from last commit
+
+2014-01-05 10:22:37 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Use the running time for container timestamps, not buffer timestamps
+	  Buffer timestamps have no real meaning here, and for selecting the next
+	  buffer we already use the running time anyway.
+
+2014-01-04 21:34:38 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/avi/gstavidemux.c:
+	  avi: use new riff api to extract nikon metadata
+	  Fixes #636143
+
+2013-11-01 16:41:43 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/inspect/plugin-rtpmanager.xml:
+	  rtprtxsend/rtprtxreceive: generate gtk doc
+
+2013-12-02 11:26:09 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  test/check: Verify rtprtxsend::ssrc-map property works as expected
+
+2013-11-29 19:35:44 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/rtpmanager/gstrtprtxreceive.h:
+	* tests/check/elements/rtpaux.c:
+	* tests/check/elements/rtprtx.c:
+	* tests/examples/rtp/client-rtpaux.c:
+	  rtprtxreceive: modify to use a payload-type map like rtprtxsend
+
+2013-11-29 19:58:26 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: do not keep history of packets with an unknown payload type
+	  This allows to disable retransmission per payload type by not putting
+	  a certain payload type in the map.
+
+2014-01-02 15:18:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	* tests/check/elements/rtpaux.c:
+	* tests/check/elements/rtpcollision.c:
+	* tests/check/elements/rtprtx.c:
+	* tests/examples/rtp/server-rtpaux.c:
+	  rtprtxsend: Allow SSRC-multiplexing and multiple payload types in the original stream
+	  Conflicts:
+	  tests/examples/rtp/server-rtpaux.c
+
+2013-11-25 15:00:45 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: Add an rtx-ssrc property to allow external control of the ssrc
+	  This is useful when one needs to know the SSRC beforehands, so that it can
+	  be used for SRTP for example.
+
+2013-11-13 15:11:35 -0500  Torrie Fischer <torrie.fischer@collabora.co.uk>
+
+	* tests/examples/rtp/.gitignore:
+	* tests/examples/rtp/Makefile.am:
+	* tests/examples/rtp/client-rtpaux.c:
+	* tests/examples/rtp/server-rtpaux.c:
+	  examples: rtp: Add end-to-end rtpbin example with RTX elements
+	  This example demonstrates how to use rtpbin with retransmission (rtx)
+	  elements set in the place of rtpbin's "aux" elements in order to
+	  enable RTP retransmission according to the rules of RFC4588.
+
+2013-11-05 17:35:01 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* docs/design/Makefile.am:
+	* docs/design/design-rtpauxiliary.txt:
+	  doc: add design-rtpauxiliary.txt to describe how rtpbin deals with auxiliary elements
+
+2014-01-02 14:48:49 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  session: also push EOS event to RTCP srcpad
+
+2014-01-02 14:46:11 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	  session: place SSRC in Retransmission event
+
+2013-11-01 16:57:15 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/rtpaux.c:
+	  tests/check: add rtpaux::test_simple_rtpbin_aux
+	  It shows how to use "set-aux-receive" and "set-aux-send"
+	  properties of rtpbin to set rtprtxsend and rtprtxreceive
+	  Build 2 pipelines, one for rtpbin as a sender and one for
+	  rtobin as a receive. Then transmit an audio stream.
+	  It also drops some packets to activate restransmission and
+	  check they are actually retransmited.
+
+2013-11-01 17:09:42 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* tests/check/elements/rtpcollision.c:
+	  tests/check: add rtpcollision::test_rtx_ssrc_collision unit test
+	  check that rtxrtpsend changes its retransmission ssrc when
+	  collision happens
+
+2013-11-06 12:34:13 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests/check: add rtprtx::test_rtxreceive_data_reconstruction
+	  This unit test verifies that retransmitted rtp packets coming out
+	  of rtprtxreceive are the same as the original ones.
+
+2013-11-05 09:33:51 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: use a realistic limit for the value of max-size-packets
+	  G_MAXINT16 is chosen because if the queue contains more than
+	  G_MAXINT16 packets, seqnum comparison will not work properly.
+
+2013-11-04 20:05:03 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	  rtprtxsend: use a GSequence to implement the buffer queue
+	  This has the advantage that searching the queue to find the
+	  buffer with the requested seqnum is done with binary search.
+
+2013-11-04 18:38:24 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	* tests/check/elements/rtprtx.c:
+	  rtprtxsend: retransmit packets in the same order as the rtx requests
+
+2013-11-02 19:56:44 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests/check: Add unit test for rtxsend's max_size_time property
+
+2013-10-29 18:27:00 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	  rtprtxsend: Handle the max_size_time property
+	  This property allows you to specify the amount of buffers
+	  to keep in the retransmission queue expressed as time (ms)
+	  instead of buffer count (which is the max_size_buffers property).
+
+2013-11-02 15:21:08 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtprtxsend.c:
+	  rtprtxsend: keep important buffer information in a private structure
+	  This is to avoid mapping a buffer every time we need to read a seqnum
+	  or a timestamp.
+
+2013-11-01 11:58:47 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtprtx.c:
+	  tests/check: Add rtprtx::test_rtxsender_packet_retention
+	  This unit test verifies that the rtxsend element correctly maintains
+	  a buffer of already transmitted rtp packets and that it can
+	  re-transmit all of them correctly on demand. It also verifies
+	  that the limit of this buffer (max-size-packets property) is respected.
+
+2013-11-01 16:22:13 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* tests/check/elements/rtprtx.c:
+	  tests/check: add rtprtx::test_drop_multiple_sender unit test
+	  Several senders / one receiver
+	  Similar than test_drop_one_sender but with multiple senders
+	  mixed through the funnel element.
+	  It drops some packets and checks that they are retransmited
+	  correctly.
+
+2013-11-01 16:21:00 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* tests/check/elements/rtprtx.c:
+	  tests/check: add rtprtx::test_drop_one_sender unit test
+	  Test for one sender / one receiver
+	  Build the pipeline
+	  videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink
+	  and drop some buffers between rtprtxsend and rtprtxreceive
+	  Then it checks that every dropped packet has been re-sent.
+	  It also checks that not too much requests has been sent.
+
+2013-11-01 16:17:51 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/rtprtx.c:
+	  tests/check: add rtprtx::test_push_forward_seq
+	  add simple unit test that manually push buffers
+	  in rtprtxsend connected to rtprtxreceive.
+	  Drops some buffers and make sure they are retransmisted.
+
+2013-11-01 15:52:03 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* gst/rtpmanager/Makefile.am:
+	* gst/rtpmanager/gstrtpmanager.c:
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/rtpmanager/gstrtprtxreceive.h:
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	  rtpmanager: add new rtprtxsend / rtprtxreceive elements
+	  The purpose of the sender RTX object is to keep a history
+	  of RTP packets up to a configurable limit (in time). It will
+	  listen for custom retransmission events from downstream. When
+	  it receives a request for retransmission, it will look up the
+	  requested seqnum in its list of stored packets. If the packet
+	  is available, it will create a RTX packet according to RFC 4588
+	  and send this as an auxiliary stream.
+	  The receiver will listen to the custom retransmission events
+	  from the downstream jitterbuffer and will remember the SSRC1
+	  of the stream and seqnum that was requested. When it sees a
+	  packet with one of the stored seqnum, it associates the SSRC2
+	  of the stream with the SSRC1 of the master stream. From then
+	  on it knows that SSRC2 is the retransmission stream of SSRC1.
+	  This algorithm is stated in RFC 4588. For this algorithm to
+	  work, RFC4588 also states that no two pending retransmission
+	  requests can exist for the same seqnum and different SSRCs or
+	  else it would be impossible to associate the retransmission with
+	  the original requester SSRC.
+	  When the RTX receiver has associated the retransmission packets,
+	  it can depayload and forward them to the source pad of the element.
+	  RTX is SSRC-multiplexed
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084
+
+2013-11-05 16:36:46 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* docs/design/Makefile.am:
+	* docs/design/design-rtpretransmission.txt:
+	  doc: add design for rtp retransmission
+	  Describe how rtprtxsend and rtprtxreceive generally work
+	  but also how the association algorithm is implemented.
+
+2014-01-02 20:23:05 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: use status code macro instead of 407
+	  Rest of the code is using the _PROXY_AUTHENTICATION_REQUIRED
+	  macro too. Easier to understand if you don't recall HTTP
+	  error codes by heart.
+
+2013-12-31 21:31:43 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* ext/shout2/gstshout2.c:
+	* ext/shout2/gstshout2.h:
+	  shout2send: change audio_format field to format
+	  This element and the underlying libshout2 library
+	  can handle video media files too. The code already
+	  handles video/webm so the name gets confusing. Also
+	  add and use DEFAULT_FORMAT macro Instead of hardwiring
+	  SHOUT_FORMAT_VORBIS at init
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721342
+
+2013-12-31 20:09:29 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2send: clarify meaning of the URL prop
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721342
+
+2013-12-27 12:27:32 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* ext/shout2/gstshout2.c:
+	  shout2send: docs, add a sample pipeline
+	  And finish adding shout2send to the docs while at it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721342
+
+2013-12-31 15:00:22 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	  gdkpixbufoverlay: remove spurious @see_also
+
+2013-12-06 17:08:54 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: support any video formats and any caps features if deinterlace mode allows it
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719636
+
+2013-12-31 13:31:52 +0100  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: Handle v4l2_ioctl() errors even in error handling
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721268
+
+2014-01-01 12:11:43 -0800  Jeremy Huddleston Sequoia <jeremyhu@apple.com>
+
+	* sys/osxvideo/Makefile.am:
+	* sys/osxvideo/osxvideosink.h:
+	* sys/osxvideo/osxvideosink.m:
+	  osxvideo: unifdef -DRUN_NS_APP_THREAD
+
+2014-01-01 12:10:01 -0800  Jeremy Huddleston Sequoia <jeremyhu@apple.com>
+
+	* sys/osxvideo/cocoawindow.m:
+	* sys/osxvideo/osxvideosink.h:
+	  osxvideo: Assume SDK and deployment target are at least Snow Leopard
+
+2014-01-01 12:23:50 -0800  Jeremy Huddleston Sequoia <jeremyhu@apple.com>
+
+	* configure.ac:
+	  configure: Disable osxvideo on Leopard and earlier
+	  This also moves the "other platforms" check in OS X video to before the
+	  variable is read
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721245
+
+2013-12-31 14:57:27 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: add AUX receiver unit test
+
+2013-12-31 13:20:01 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: improve rtpbin test
+
+2013-12-31 13:16:46 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: add some docs about AUX elements
+
+2013-12-31 13:01:22 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: add AUX sender unit test
+
+2013-12-31 12:31:25 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	  rtpbin: add support for AUX sender and receiver
+	  AUX elements are elements that can be inserted into the rtpbin
+	  pipeline right before or after 1 or more session elements.
+	  The AUX elements are essential for implementing functionality such
+	  as error correction (FEC) and retransmission (RTX).
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
+
+2013-12-31 12:22:39 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: add decoder test
+
+2013-12-30 17:36:42 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: make request_element method internally
+	  We can use the same method to create encoder and decoder elements, they
+	  are just internal elements that we create.
+
+2013-12-31 10:25:28 +0100  Stéphane Cerveau <scerveau@gmail.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: Skip id3 tag
+	  Skip id3 tag during wav parse.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721241
+
+2013-12-31 10:10:05 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* sys/osxaudio/gstosxcoreaudio.h:
+	* sys/osxvideo/cocoawindow.m:
+	* sys/osxvideo/osxvideosink.h:
+	  osx: Make OSX version checks more consistent
+	  And especially also consider update versions, e.g. 10.5 with updates
+	  will be 1051 or similar and thus bigger than MAC_OS_X_VERSION_10_5 but
+	  still won't have the API we want to use.
+
+2013-12-31 10:07:22 +0100  Jeremy Huddleston <jeremyhu@freedesktop.org>
+
+	* sys/osxvideo/osxvideosink.h:
+	  osxvideosink: Fix build on updated OS X Leopard
+	  https://bugzilla.gnome.org/show_bug.cgi?id=721245
+
+2013-12-30 17:23:22 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/avi/gstavimux.c:
+	  avimux: Add missing break
+	  I guess no-one noticed we no longer could mux WMV3 ...
+	  COVERITY CID 1139759
+
+2013-12-30 17:20:37 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpvrawpay.c:
+	  rtpvrawpay: Add missing break
+	  COVERITY CID 1139762
+
+2013-12-30 17:00:45 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: internal-ssrc is no longer deprecated
+
+2013-12-30 16:59:20 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: add Since tags
+
+2013-12-30 16:52:28 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	  rtpbin: add signal for new jitterbuffer
+	  Emit a signal when a new jitterbuffer is created so that the app can
+	  have a chance to configure it.
+
+2013-12-30 16:28:57 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* tests/check/elements/rtpbin.c:
+	  rtpbin: handle multiple encoder instances
+	  Keep track of elements that are added to multiple sessions and make sure
+	  we only add them to the rtpbin once and that we clean them when no
+	  session refers to them anymore.
+
+2013-12-30 15:16:09 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: add unit test for encoder element
+
+2013-12-30 15:15:43 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: fix memory leaks
+
+2013-12-30 15:03:34 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: fix leak
+
+2013-12-30 15:00:50 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: expect the pads on the encoders
+	  Don't use request pads for the encoder elements, the signal handler
+	  should request the pads and make sure they are available with the right
+	  name.
+
+2013-12-30 14:56:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	  rtpbin: request-rtp-encoder are no action signals
+	  The request-rtp-encoder signals are not action signals so mark them
+	  correctly and use an accumulator to collect the result value.
+
+2013-12-30 14:36:45 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: emit midi-base-note tag from data in 'smpl' chunk
+	  Add parsing of the 'smpl' chunk. Right now we only grab the midi-base-note and
+	  emit it as a tag.
+
+2013-12-26 12:05:19 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
+	  When a collision is found on the internal ssrc, we have to change it.
+	  Ideally, we want also the payloader upstream to follow this change and use
+	  the new internal ssrc. Ideally we want this condition to be always met:
+	  if there is one payloader sending on this session, its ssrc should match the
+	  internal ssrc.
+
+2013-12-26 11:04:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: allow setting internal-ssrc again
+
+2013-12-30 13:31:45 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/y4m/gsty4mencode.c:
+	  y4mencode: Remove dead code
+	  set/get property isn't used
+
+2013-12-30 13:30:24 +0100  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpqcelpdepay.c:
+	  rtpqcelpdepay: Remove uneeded variable
+
+2013-12-05 15:53:52 -0800  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	  rtpbin: allow dynamic RTP/RTCP encoders/decoders
+	  * gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
+	  added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
+	  and request-rtcp-decoder). The user will be able to provide encoders
+	  or decoders dynamically. The encoders must follow the srtpenc API and
+	  the decoders the srtpdec API. Having separate signals for RTP and RTCP
+	  allows the user to use different encoders/decoders or provide the same
+	  one (e.g. that would be the case for srtpenc).
+	  Also, rtpbin now allows application/x-srtp in its pads.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719938
+
+2013-12-27 16:51:32 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: dynamically recalculate RTX parameters
+	  Use the round-trip-time and average jitter to dynamically calculate the
+	  retransmission interval and expected packet arrival time.
+	  Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
+
+2013-12-27 16:50:52 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: calculate average jitter
+
+2013-12-27 16:48:48 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpsession: use RTT from the Retransmission event
+	  Place the estimated RTT in the Retransmission event and let the session
+	  manager use that instead of the hardcoded value.
+
+2013-12-27 15:57:39 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: take more accurate running-time for NACK
+	  Don't use the current time calculated from the tmieout loop for when we
+	  last scheduled the NACK because it might be unscheduled because of a max
+	  packet misorder and then we don't accurately calculate the current time.
+	  Instead, take the current element running time using the clock.
+
+2013-12-30 11:06:38 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* tests/check/elements/wavpackdec.c:
+	  wavpackdec: Send a CAPS event in the unit test
+
+2013-12-27 02:14:02 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: improve mss_mode/fragmented special handling
+	  Make it clear what should be handled purely by mss mode:
+	  1) Expose the streams on the first moof as there are no moov atoms
+	  2) Properly cleanup streams on flushes
+	  Add a note about the meaning of upstream_newsegment and mss_mode
+	  for future reference.
+	  Make all other special fragment handling shared for both dash
+	  and mss streams.
+
+2013-12-12 10:50:27 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: drain the adapter before pushing EOS
+	  In a fragmented scenario, qtdemux is operating in push mode
+	  and it gets a fragmented buffer. While processing its data
+	  downstream gets unlinked (or a input-selector changes its
+	  active pad and returns not-linked). Qtdemux stops processing
+	  this fragment and returns not-linked upstream, leaving the
+	  remaining data in its adapter.
+	  When it gets an EOS it should make sure that all the data it
+	  had received is pushed before pushing EOS.
+
+2013-12-26 23:21:47 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2send: drop IP only requirement for _set_host()
+	  libshout2 (we require > 2.0 at config time) supports
+	  both IP and hostname for _set_host(). Dropped an
+	  outdated FIXME regarding this limitation, adjusted
+	  some comments and changed the param blurb to reflect
+	  this too.
+
+2013-12-26 21:43:34 -0300  Reynaldo H. Verdejo Pinochet <r.verdejo@sisa.samsung.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2send: Retarget FIXME to 2.0
+
+2013-12-26 11:21:36 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: use aggregate control for PLAY/PAUSE/TEARDOWN
+	  Use the aggregate control instead of the original request url to perform
+	  PAUSE/PLAY and TEARDOWN.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=721003
+
+2013-12-24 14:40:25 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/debugutils/rndbuffersize.c:
+	  rndbuffersize: Proxy CAPS, ALLOCATION, SCHEDULING and srcpad events properly
+
+2013-12-24 00:43:39 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: adpcm max block align is 8192
+
+2013-12-23 12:23:27 -0600  Brendan Long <b.long@cablelabs.com>
+
+	* configure.ac:
+	  vp9dec: Require vpx >= 1.3.0 for building vp9dec and vp9enc
+	  Previous versions did not have a stable bitstream for VP9.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720986
+
+2013-12-23 15:46:48 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Use correct codec id for ADPCM/DVI
+
+2013-12-23 15:44:30 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Check for the correct size of codec_data in the ACM case
+
+2012-01-14 19:58:17 +0100  Nicola Murino <nicola.murino@gmail.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: basic adpcm support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=664339
+
+2013-12-20 11:45:38 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/isomp4/descriptors.c:
+	  qtdemux: Fix calcuation of descriptor length
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720813
+
+2013-12-22 22:33:39 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* autogen.sh:
+	* common:
+	  Automatic update of common submodule
+	  From dbedaa0 to d48bed3
+
+2013-12-22 21:56:03 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* po/Makevars:
+	  po: set gettext domain in Makevars so we don't have to patch the generated Makefile.in.in
+	  https://bugzilla.gnome.org/show_bug.cgi?id=705455
+
+2013-12-19 16:50:10 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: on receive error only unmap and unref buffer if one was alloced and mapped
+	  coverity CID 1139866.
+
+2013-12-19 12:47:22 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiudpsink: fix misleading comment
+	  Those are not allocated on the stack.
+
+2013-12-17 18:28:25 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  vpx: Mark VP9 support as non-experimental
+	  There was a libvpx release with VP9 support now and the bitstream
+	  is frozen too.
+
+2013-12-15 21:04:11 -0800  Todd Agulnick <todd@agulnick.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  Some compiler warning fixes to satisfy XCode compiler
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720513
+
+2013-12-16 16:17:07 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/taglib/gstid3v2mux.cc:
+	  id3v2mux: Set picture type in the APIC frames
+
+2013-12-16 16:14:52 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/taglib/gstid3v2mux.cc:
+	  id3v2mux: Set image-description from the info struct, not the caps
+
+2013-12-16 10:02:37 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstwavpackparse.c:
+	* gst/audioparsers/gstwavpackparse.h:
+	  wavpackparse: Post AUDIO_CODEC tag
+
+2013-12-16 10:00:37 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstsbcparse.h:
+	  sbcparse: Post AUDIO_CODEC tag
+
+2013-12-16 09:58:31 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstflacparse.h:
+	  flacparse: Post AUDIO_CODEC tag
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720512
+
+2013-12-16 09:56:29 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstdcaparse.h:
+	  dcaparse: Post AUDIO_CODEC tag
+
+2013-12-16 09:54:38 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstamrparse.c:
+	* gst/audioparsers/gstamrparse.h:
+	  amrparse: Post AUDIO_CODEC tag
+
+2013-12-16 09:49:48 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstac3parse.h:
+	  ac3parse: Post AUDIO_CODEC tag
+
+2013-12-16 09:46:16 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstaacparse.h:
+	  aacparse: Post AUDIO_CODEC tag
+
+2013-12-16 09:41:14 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audioparsers/gstmpegaudioparse.c:
+	  mpegaudioparse: Use pbutils functionality to create the AUDIO_CODEC tag
+
+2013-12-13 17:36:36 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Add error message if the app tries to set the internal-ssrc
+
+2013-12-13 16:08:35 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Only count nacks when a nack packet is received
+	  Not when any RTCP feedback packet is.
+
+2013-12-12 23:22:41 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* tests/check/elements/rtpcollision.c:
+	  tests: Initialize segment in rtpcollision test
+
+2013-12-13 15:57:36 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Process PSFB FIR requests which lack the media ssrc
+	  According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
+	  have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
+	  So in that case, we ignore the retained feedback and just let it through
+	  to the rtp_session_process_fir() function which will check for the actual
+	  SSRC inside the FCI.
+	  Fixes a regression introduced by commit 57c27ec3
+
+2013-11-14 16:19:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	  rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
+	  Previously, when the session had multiple internal sender SSRCs, it would
+	  issue SR reports with RB blocks only on the first RTCP timeout and afterwards
+	  SR reports would be sent empty. This was because the "generation" number
+	  in RTPSource would increase more than once during the same cycle and afterwards
+	  it would always be greater than the session's generation, which would cause
+	  it to be skipped from being included in RBs.
+	  This commit fixes this problem by:
+	  1) Increasing the RTPSource generation only at the end of each cycle,
+	  which essentially fixes the problem but only when the internal senders
+	  are less than GST_RTCP_MAX_RB_COUNT.
+	  2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
+	  SR the given RTPSource has been reported in, which also fixes the problem
+	  when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
+	  necessary because of the fact that any RTPSource is marked as reported
+	  in itself's SR and makes it impossible to know if it has been reported
+	  in other SRs too or not, and which.
+
+2013-11-14 16:23:35 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* tests/check/elements/rtpsession.c:
+	  tests/check: add an rtpsession unit test to verify all RBs are included in all SRs, roundrobin
+	  This test checks that when we have multiple internal sender sources
+	  in rtpsession, SRs contain RBs for every other sender source, and that
+	  they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT,
+	  which is the max number of RBs that can fit in a SR.
+
+2013-12-12 16:01:10 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* docs/design/design-rtpcollision.txt:
+	  docs: improve docs
+
+2013-11-05 18:03:48 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* docs/design/Makefile.am:
+	* docs/design/design-rtpcollision.txt:
+	  doc: add design-rtpcollision.txt that explains when GstRTPCollision is created
+	  It also talks about "BYE only the corresponding source, not the whole session."
+
+2013-11-05 12:31:54 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* tests/check/elements/rtpcollision.c:
+	  tests/check: improve rtpcollision::test_master_ssrc_collision to ensure that a collision does not BYE the whole session
+	  Conflicts:
+	  tests/check/elements/rtpcollision.c
+
+2013-11-01 17:07:57 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/rtpcollision.c:
+	  tests/check: add rtpcollision::test_master_ssrc_collision unit test
+	  It checks the payloader changes its ssrc when collision happens
+
+2013-12-12 10:38:43 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	  rtpsession: keep extra stats for scheduling BYE
+	  Keep an extra stats structure for scheduling the BYE packets. When we
+	  decide to schedule BYE, make a copy of the current stats into the
+	  bye_stats. Then while we schedule the BYE, update and use only the
+	  bye_stats. When we finished scheduling the BYE packet, we use the
+	  regular stats again.
+
+2013-12-12 10:34:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: when we schedule BYE, only deal with BYE sources
+	  When we are doing the RTCP timeout to schedule BYE packets, don't
+	  generate RTCP for all sources but only for the sources marked as BYE.
+
+2013-12-12 10:32:48 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: reset state after scheduling BYE
+	  After we do RTCP, we are not scheduling bye anymore.
+
+2013-12-12 10:31:38 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: also count NACKS when no signal was pending
+
+2013-12-12 10:09:25 +0100  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  session: ignore RTCP packets for the BYE sources
+	  When we are scheduling BYE packets, ignore all RTCP for the sources that
+	  are scheduling a BYE packet. Other sources that are not scheduling BYE
+	  should continue receiving RTCP packets as usual.
+
+2013-11-04 11:48:21 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	  rtpsession: determine if the session is doing point-to-point
+	  In this case T_dither_max is set to 0 according to RFC 4585
+
+2013-12-10 11:57:37 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: serialize events in the buffer
+	  Serialize events into the jitterbuffer by inserting them with a -1
+	  seqnum.
+	  Update unit test to expect events from the streaming thread.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986
+
+2013-12-10 11:04:06 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: detect -1 seqnum
+	  Keep the seqnum as a full guint so that we can check for -1 entries and
+	  deal with them correctly.
+	  Immediately try to push -1 seqnum.
+
+2013-12-10 11:01:03 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: reorganize jitterbuffer items
+	  Keep the oldest item at the head and the newest items on the tail. This
+	  makes it easier to deal with -1 seqnums.
+
+2013-12-09 23:34:10 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  jitterbuffer: correctly check for invalid values
+	  Check for -1 on the guint from the buffer item instead of on the guint16
+	  or guint32.
+	  Also insert -1 seqnum at the head of the jitterbuffer.
+
+2013-12-08 16:49:55 +0100  Alessandro Decina <alessandro.d@gmail.com>
+
+	* sys/osxvideo/cocoawindow.m:
+	* sys/osxvideo/osxvideosink.m:
+	  osxvideosink: fix segfault when dealing with padded frames
+	  Fixes crashes with vtdec ! osxvideosink where VideoToolbox outputs padded UYVY
+
+2013-12-05 12:15:29 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/law/mulaw-decode.c:
+	  mulawdec: Require caps to be set before accepting any data
+
+2013-12-05 12:15:19 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/wavpack/gstwavpackdec.c:
+	  wavpackdec: Require caps to be set before accepting any data
+
+2013-12-05 12:13:33 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/speex/gstspeexdec.c:
+	  speexdec: Require caps to be set before accepting any data
+
+2013-12-05 12:13:10 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: Require caps to be set before accepting any data
+
+2013-12-05 11:42:15 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp9dec.c:
+	  vpx: Use new gst_video_decoder_set_needs_format() API
+
+2013-12-04 16:23:43 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: Free device_info in accepts caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719811
+
+2013-12-04 21:57:48 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtptheorapay.c:
+	  rtptheorapay: Don't send headers twice if we got them from the caps already
+
+2013-12-04 21:57:04 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtptheorapay.c:
+	  rtptheorapay: Don't leak config data when receiving a second CAPS event
+
+2013-12-04 21:55:53 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtpvorbispay: Don't send headers twice if we got them from the caps already
+
+2013-12-04 21:54:16 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtpvorbispay: Don't leak config data when receiving a second CAPS event
+
+2013-12-04 21:17:03 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpstreamdepay.c:
+	* gst/rtp/gstrtpstreamdepay.h:
+	  rtpstreamdepay: Add RFC4571 RTP stream depayloading element
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719829
+
+2013-12-04 10:12:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpstreampay.c:
+	* gst/rtp/gstrtpstreampay.h:
+	  rtpstreampay: Add RFC4571 RTP stream payloading element
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719829
+
+2013-12-03 15:08:25 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: improve fragment-start tracking
+	  Some buffers can have multiple moov atoms inside and the strategy
+	  of using the gst_adapter_prev_pts timestamp to get the base timestamp
+	  for the media of the fragment would fail as it would reuse the same
+	  base timestamp for all moofs in the buffer instead of accumulating
+	  the durations for all of them.
+	  Heres a better explanation of the issue:
+	  qtdemux receives a buffer where PTS(buf) = X
+	  buf -> moofA | moofB | moofC
+	  The problem was that PTS(buf) was used as the base timestamp for
+	  all 3 moofs, causing all buffers to be X based. In this case we want
+	  only moofA to be X based as it is what the PTS on buf means, and the
+	  other moofB and moofC just use the accumulated timestamp from the
+	  previous moofs durations.
+	  To solve this, this patch uses gst_adapter_prev_pts distance
+	  result, this allows qtdemux to calculate if it should use the
+	  resulting pts or just accumulate the samples as it can identify
+	  if the moofs belong to the same upstream buffer or not.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719783
+
+2013-11-21 12:29:28 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: add support for multi-planar V4l2 API in DMABUF mode
+	  Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754
+
+2013-11-19 17:16:27 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	  v4l2: refactor by emulating one v4l2_plane in non-MPLANE mode
+	  so that the buffer informations can be retrieved the same way
+	  in both MPLANE and non-MPLANE mode.
+	  Here "emulating" means "manually fill in the plane".
+	  Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754
+
+2013-11-13 12:05:40 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: add support for multi-planar V4L2 API
+	  This api is in linux kernel since version 2.6.39,
+	  and present in all version 3.
+	  The commit that adds the API in master branch of the
+	  linux kernel source is:
+	  https://github.com/torvalds/linux/commit/f8f3914cf922f5f9e1d60e9e10f6fb92742907ad
+	  v4l2 doc: "Some devices require data for each input
+	  or output video frame to be placed in discontiguous
+	  memory buffers"
+	  There are newer structures 'struct v4l2_pix_format_mplane'
+	  and 'struct v4l2_plane'.
+	  So the pixel format is not setup with the same API when using
+	  multi-planar.
+	  Also for gst-v4l2, one of the difference is that in GstV4l2Meta
+	  there are now one mem pointer for each maped plane.
+	  When not using multi-planar, this commit takes care of keeping
+	  the same code path than previously. So that the 2 cases are
+	  in two different blocks triggered from V4L2_TYPE_IS_MULTIPLANAR.
+	  Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=712754
+
+2013-12-04 09:12:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	  audioparsers: don't leak template caps
+
+2013-12-03 21:41:28 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstamrparse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	* tests/check/elements/aacparse.c:
+	  audioparsers: use ACCEPT_INTERSECT flag
+	  The parser can accept input that is not completely specified. Use the
+	  ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
+	  check for intersection only. This allows us to proxy downstream
+	  constraints while still allowing non-subset caps as input.
+	  We can then also remove the appended template caps workaround.
+	  Make a unit-test to check the new feature.
+	  This reverts commit 26040ee38cb9e7c42f3d9a0282b3e5cace7ca42d
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
+
+2013-12-03 21:36:54 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	  audioparsers: remove fields from filter
+	  We need to remove the fields from the filter when we can convert
+	  between them.
+
+2013-12-03 21:29:13 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	  audioparsers: refactor code to remove caps fields
+
+2013-12-02 00:10:43 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: microoptimisation: avoid some unnecessary GValue copies
+
+2013-12-01 23:32:20 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: fix off-by-one crash when downstream caps contain a list of framerates
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719544
+
+2013-11-29 11:26:05 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Use the timestamp of the moof as the base fragment start
+	  In SmoothStreaming fragmented scenario, the timestamps are calculated
+	  starting from the fragment buffer timestamp. When there is a not-linked
+	  return from downstream, qtdemux will return upstream and will keep the
+	  non-pushed data into its adapter.
+	  On a new fragment buffer pushed to qtdemux, the new buffer timestamp
+	  would overwrite the previous one that should be used on the still
+	  to be pushed buffers. Because of this, this patch will also
+	  update the fragment_start timestamp from the adapter last pts
+	  to make sure the moof and timestamps are in sync and will result
+	  in correct timestamps for all fragments.
+
+2013-11-15 08:54:07 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: avoid re-reading the same moov and entering into loop
+	  In the scenario of "mdat | moov (with fragmented artifacts)" qtdemux
+	  could read the moov again after the mdat because it was considering the
+	  media as a fragmented one.
+	  To avoid this loop this patch makes it store
+	  the last processed moov_offset to avoid parsing it again.
+	  And it also checks if there are any samples to play before
+	  resturning to the mdat, so that it knows there is new data to be played.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=691570
+
+2013-11-15 00:52:53 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: do not free streams if they were not created locally
+	  When parsing a trak only free streams on failures if those streams
+	  were created locally. They could have been created from a previous
+	  fragment, in this case we they have valid info from the other fragment.
+	  Including pads.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=691570
+
+2013-11-29 19:57:46 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videomixer/blend.c:
+	  videomixer: Simplify NV12/21 blending code macros
+
+2013-11-29 19:50:24 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/videomixer/blend.c:
+	  videomixer: Fix segfault when filling the background of a UYVY frame
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712401
+
+2013-11-29 09:21:52 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix compilation with gst debuging disabled
+	  qtdemux.c:9452:1: error: label at end of compound statement
+
+2013-11-27 17:02:00 +0100  Jonas Holmberg <jonashg@axis.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	  rtph264pay: Map inbuffer once only
+	  Do not call gst_buffer_extract() twice since each call will map and
+	  unmap the biffer.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719434
+
+2013-11-28 11:58:42 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* tests/check/elements/videofilter.c:
+	  videoflip: Add unit test for the 'automatic' method
+	  These new tests send a tag event before seding the buffer. Tested case are an
+	  empty tag list, a tag list with orientation-180 set and an invalid orientation value.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719497
+
+2013-11-28 16:09:04 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/videofilter/gstvideoflip.c:
+	  videoflip: don't crash on tag events without orientation tag
+	  Would crash in g_free() trying to free an uninitialised pointer.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=719497
+
+2013-11-28 16:50:42 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: don't unref buffer twice
+	  Cleaning the packet info will already unref the buffer.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078
+
+2013-11-28 22:35:02 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Add HydrogenAudio ReplayGain tags
+	  Identical to the itunes (tm) version, but labelled with
+	  org.hydrogenaudio.replaygain as the producer.
+
+2013-11-27 16:15:12 +0100  Mathieu Duponchelle <mduponchelle1@gmail.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: explicitly fail when alpha information would have been lost.
+
+2013-05-29 16:06:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* .gitignore:
+	  gitignore: Updated to ignore *.swp and .dirstamp
+
+2013-11-26 11:17:42 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: Allow a bit more variation when detecting common framerates
+	  Instead of +/- 1ns we allow 2ns now. Due to rounding errors there are
+	  some Matroska files out there with 33.333331ms per frame for 30fps.
+
+2013-11-26 10:20:31 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: Use gst_util_double_to_fraction() instead of GValue magic
+
+2013-11-25 14:03:21 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/videofilter/gstvideoflip.c:
+	  videoflip: Set default method at contruction
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712333
+
+2013-05-29 15:57:09 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Use space instead of tabs
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712754
+
+2013-05-29 15:44:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Fix header indentation so it's readable again
+	  It's unfortunate to have to do this, but with the mix of tabs and space, plus all the random
+	  indentation this header has become very hard to read.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712754
+
+2013-11-25 17:38:06 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* tests/check/elements/rtpjitterbuffer.c:
+	  check: fix jitterbuffer check
+	  Don't advance the clock to 240ms too early.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710013
+
+2013-11-25 11:45:33 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: deprecate max-errors
+	  The property wasn't use internally, let the base class handle the
+	  number of errors to tolerate.
+
+2013-11-25 15:49:07 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: improve clear-pt-map handling
+	  Don't reset the expected output seqnum when clearing the pt map because this
+	  could stall the jitterbuffer forever.
+	  Add a unit test for this.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
+
+2013-10-28 21:33:22 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: let the base class decide when to return an error
+	  The base videodecoder class has an error counting feature to tolerate
+	  a few errors before posting an error message. So don't force the
+	  error and let the base class decide when it should happen
+	  https://bugzilla.gnome.org/show_bug.cgi?id=710762
+
+2013-10-28 21:28:33 -0300  Thiago Santos <ts.santos@sisa.samsung.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: Add data skipping on input
+	  Add missing bytes skipping when bad input is received.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=710762
+
+2013-11-25 12:13:43 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Discard 2 byte subpicture packets
+	  As for text subtitles and as suggested in #712643, throw
+	  away the 2 byte terminator packets that some encoders insert.
+	  This will make things better when remuxing and causes generation
+	  of gap events.
+
+2013-11-25 00:34:21 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: fix wake-up when new buffers come in after running empty
+	  Spotted by 'gratias' on IRC. Probably introduced in recent refactoring.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=715039
+
+2013-11-23 12:15:40 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: correctly handle negative relative timestamps
+	  ... rather than scaling these as unsigned.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712744
+	  Based on patch by Krzysztof Kotlenga <pocek@users.sf.net>
+
+2013-09-14 03:27:09 +0200  MathieuDuponchelle <mathieu.duponchelle@epitech.eu>
+
+	* gst/videomixer/videomixer2.c:
+	* gst/videomixer/videomixer2.h:
+	  videomixer2: Merge tag events to send them in collected.
+	  Otherwise there were race conditions where we would send tags
+	  on a flushing srcpad.
+	  We have a test for that in GES, but this should be tested
+	  systematically with harness in the future as I believe it
+	  is useful for exactly that kind of cases.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=708165
+
+2013-11-14 17:29:50 -0300  Thibault Saunier <thibault.saunier@collabora.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Use GstVideoInfo helper to create caps for raw video
+	  This way we do not miss mandatory fields in caps.
+	  At the same time use the gst_pb_utils_get_codec_description
+	  helper to get codec description.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712335
+
+2013-11-14 16:11:38 -0300  Thibault Saunier <thibault.saunier@collabora.com>
+
+	* gst/matroska/Makefile.am:
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Use GstVideoInfo helper to create caps for raw video
+	  This way we do not miss mandatory fields in caps.
+	  At the same time use the gst_pb_utils_get_codec_description helper to
+	  get codec description.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712328
+
+2013-11-13 20:18:17 -0300  Thibault Saunier <thibault.saunier@collabora.com>
+
+	* gst/multifile/gstmultifilesrc.c:
+	* gst/multifile/gstmultifilesrc.h:
+	  multifilesrc: Implement seeking in case of multiple images
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712254
+
+2013-11-22 12:26:21 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: pass downstream flowreturn to upstream
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712722
+
+2013-11-18 14:27:48 +0100  Michael Olbrich <m.olbrich@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: clear cached caps on close
+	  A different device with different caps may be used for the next open.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712611
+
+2013-11-21 15:30:34 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/wavpack/gstwavpackcommon.c:
+	* ext/wavpack/gstwavpackstreamreader.c:
+	* gst/apetag/gstapedemux.c:
+	* gst/autodetect/gstautoaudiosink.c:
+	* gst/autodetect/gstautoaudiosrc.c:
+	* gst/autodetect/gstautovideosink.c:
+	* gst/autodetect/gstautovideosrc.c:
+	* gst/dtmf/gstrtpdtmfsrc.c:
+	* gst/isomp4/atoms.c:
+	* gst/matroska/matroska-demux.c:
+	  g_memmove() is deprecated
+	  Just use plain memmove(), g_memmove() is deprecated in
+	  recent GLib versions.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712811
+
+2013-11-21 11:32:15 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtp/gstrtpvorbisdepay.c:
+	* gst/rtp/gstrtpvorbispay.c:
+	  rtpvorbisdepay: handle packets > 0xffff
+	  Handle input packet sizes larger than 16 bits in the depayloader.
+	  Remove size restrictions on the payloader.
+
+2013-11-21 11:30:28 +0100  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtp/gstrtptheoradepay.c:
+	* gst/rtp/gstrtptheorapay.c:
+	  rtptheoradepay: handle packets > 0xffff
+	  Reorganize some things in the depayloader so that it can handle packets larger
+	  than 16 bits.
+	  Remove the size restriction on the payloader.
+
+2013-11-21 02:28:27 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_dump.c:
+	* gst/isomp4/qtdemux_types.c:
+	  isomp4: Handle mp4s subpicture streams better.
+	  Clean up the handling of mp4s streams. Use the generic esds
+	  descriptor function to extract the palette, instead of hard coding
+	  a wrong magic offset.
+	  Add some more size safety checks when parsing ES descriptors, and
+	  replace magic numbers with the descriptive constants that are already
+	  defined.
+	  Enhance dump output for stsd atoms.
+	  Streams from both bug 712643 and historic bug 568278 now both work
+	  correctly.
+	  Fixes: #712643
+
+2013-11-20 22:08:25 +1100  Jan Schmidt <thaytan@noraisin.net>
+
+	* gst/isomp4/fourcc.h:
+	  qtdemux: Sort fourcc declarations and remove duplicates
+
+2013-11-20 21:41:47 +1100  Jan Schmidt <thaytan@noraisin.net>
+
+	* gst/isomp4/Makefile.am:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/ftypcc.h:
+	* gst/isomp4/gstqtmuxmap.c:
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_dump.c:
+	* gst/isomp4/qtdemux_fourcc.h:
+	* gst/isomp4/qtdemux_types.c:
+	  qtdemux: Merge all the fourcc headers into one
+	  Remove qtdemux_fourcc.h and ftypcc.h and put it all in fourcc.h
+
+2013-11-19 10:10:51 +0100  Wim Taymans <wim.taymans@gmail.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: avoid mapping the buffer
+	  Reuse the parsed structure to get the timestamps.
+
+2013-11-18 17:13:49 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: fix 'make check'
+	  Fix generic/states check. Also, g_return_if_fail() is
+	  not for internal state checking.
+
+2013-11-18 14:44:36 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
+	* ext/jack/gstjackaudiosink.c:
+	* ext/jack/gstjackaudiosrc.c:
+	* ext/jpeg/gstjpegdec.c:
+	* ext/pulse/pulsesink.c:
+	* ext/pulse/pulsesrc.c:
+	* ext/raw1394/gstdv1394src.c:
+	* ext/raw1394/gsthdv1394src.c:
+	* gst/audiofx/audioecho.c:
+	* gst/audiofx/audiofxbasefirfilter.c:
+	* gst/audiofx/audiopanorama.c:
+	* gst/autodetect/gstautoaudiosink.c:
+	* gst/autodetect/gstautoaudiosrc.c:
+	* gst/autodetect/gstautovideosink.c:
+	* gst/autodetect/gstautovideosrc.c:
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/flv/gstflvmux.c:
+	* gst/multifile/gstmultifilesink.c:
+	* gst/multifile/gstmultifilesink.h:
+	* gst/multifile/gstsplitfilesrc.c:
+	* gst/multipart/multipartdemux.c:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/smpte/gstsmptealpha.c:
+	* gst/udp/gstmultiudpsink.c:
+	* gst/videobox/gstvideobox.c:
+	* gst/wavparse/gstwavparse.c:
+	* sys/oss4/oss4-sink.c:
+	* sys/oss4/oss4-source.c:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/ximage/gstximagesrc.c:
+	  docs: get rid of 'Since: 0.10.x' markers
+	  And some gtk-doc markup fixes.
+
+2013-11-16 12:15:14 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	  rtpmanager: fix Since markers
+	  Should be next stable release series version
+
+2013-11-15 13:48:07 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix stats property field names and documentation
+
+2013-11-15 15:20:14 +0100  Torrie Fischer <torrie.fischer@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpstats.c:
+	* gst/rtpmanager/rtpstats.h:
+	  gstrtpsession: Implement a number of feedback packet statistics
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693
+
+2013-11-13 17:11:08 -0300  Thiago Santos <ts.santos@partner.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: remove math operation from loop
+	  The elst_offset doesn't change inside the loop, so compute it
+	  outside
+
+2013-11-14 20:54:32 +0100  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/isomp4/qtdemux.c:
+	  qtmux: fix playback regression
+	  In ae1150e85cf99d7482933aa6f7e4f012fe45a3ec flipping a condition misaligned the
+	  else branch, where for there condition that was change there is none.
+	  Fixes #712303
+
+2013-11-14 09:20:06 +0100  Wim Taymans <wim.taymans@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: rename property to 'stats'
+	  This makes the unit test work.
+	  We can later also add more stats, not specific to retransmission.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711411
+
+2013-11-12 11:19:25 -0500  Torrie Fischer <torrie.fischer@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: implement rtx statistics
+
+2013-11-13 10:42:21 +0000  Marc Leeman <marc.leeman@gmail.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: print FOURCC_FORMAT when enumerating
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712206
+
+2013-11-06 12:40:06 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: advance expected seqnum after dropping
+	  After dropping a buffer, move our expected seqnum
+	  Conflicts:
+	  gst/rtpmanager/gstrtpjitterbuffer.c
+
+2013-11-04 15:46:22 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpgstpay.c:
+	  gstpay: only send one caps
+	  Only send one caps in a packet. Two caps can happen when setcaps is called and
+	  the config-interval expires at the same time.
+
+2013-11-13 10:23:19 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: Use the synced buffer mode in auto mode if a clock provider is in the SDP
+
+2013-11-08 11:09:21 +0000  Marc Leeman <marc.leeman@gmail.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2: init v4l2_buffer to 0x0 before ioctl
+	  https://bugzilla.gnome.org/show_bug.cgi?id=712137
+
+2013-11-11 15:27:18 +0100  Wim Taymans <wim.taymans@gmail.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpsession: remove collision reconfigure event
+	  Remove bogus reconfigure event on collision, we don't want to send the event on
+	  the receiving RTP pad and the collision event is now handling this
+	  case.
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=711560
+
+2013-11-01 17:04:28 +0000  Julien Isorce <julien.isorce@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad
+	  See https://bugzilla.gnome.org/show_bug.cgi?id=711560
+
+2013-11-11 14:25:51 +0100  Wim Taymans <wim.taymans@gmail.com>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/rtpsession.c:
+	  check: add rtpsession test
+	  Add a basic rtpsession test to ensure that RR blocks are generated when
+	  multiple SSRC senders are active.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711270
+
+2013-11-11 13:17:25 +0100  Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+	* gst/audioparsers/gstac3parse.c:
+	  ac3parse: correctly handle timestamps when parsing x-private1-ac3
+	  ... the way it has always worked fine in a52dec.
+
+2013-11-05 10:48:33 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: fix crash when do-retransmission=true and a lot of buffers are lost
+	  The problem here was that the jitterbuffer lock was unlocked to push
+	  the event, but that caused another thread to remove the timer currently
+	  being processed, probably because the amount of rtx events
+	  (and therefore timers) was getting too high. The solution is to
+	  unlock and push the event only after timer processing has finished.
+	  fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131
+
+2013-10-24 13:16:42 +0200  Per x Johansson <perxjoh@axis.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Avoid division by zero assert in gst_matroska_demux_search_pos
+	  https://bugzilla.gnome.org/show_bug.cgi?id=711829
+
+2013-11-08 17:59:24 +0100  Philippe Normand <philn@igalia.com>
+
+	* gst/wavenc/gstwavenc.c:
+	  wavenc: generate a non-empty data header
+	  Restore the behavior of the element to the state before commit
+	  db29522a430e44450415ca3676abd1b77ee923d9. A non-empty header is
+	  generated and when the EOS event is received the header is generated
+	  again, this time with the correct size.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=711699
+
+2013-11-07 16:17:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: update receiver stats for sender
+	  An internal sender in a session is also a receiver of its own packets so update
+	  the receiver stats. Other senders in the session will use this info to generate
+	  correct RB blocks in their SR reports.
+
+2013-11-07 16:13:16 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: refactor receiver stats update
+
+2013-10-25 18:22:00 -0300  Thiago Santos <ts.santos@partner.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: handle fragmented files with mdat before moofs
+	  Assume a file with atoms in the following order: moov, mdat, moof,
+	  mdat, moof ...
+	  The first moov usually doesn't contain any sample entries atoms (or
+	  they are all set to 0 length), because the real samples are signaled
+	  at the moofs. In push mode, qtdemux parses the moov and then finds the mdat,
+	  but then it has 0 entries and assumes it is EOS.
+	  This patch makes it continue parsing in case it is a fragmented file so that
+	  it might find the moofs and play the media.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=710623
+
+2013-10-25 11:42:37 -0300  Thiago Santos <ts.santos@partner.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux.h:
+	  qtdemux: When using a buffered mdat, store all received data for later use
+	  In push mode, when qtdemux can't use a seek to skip the mdat buffer it has
+	  to buffer it for later use.
+	  The issue is that after parsing the next moov/moof, there might be some
+	  trailing bytes from the next atom in the file. This data was being discarded
+	  along with the already parsed moov/moof and playback would fail to continue
+	  after the contents of this moov/moof are played.
+	  This is particularly bad on fragmented files that have the mdat before the
+	  corresponding moof. So you'd get:
+	  mdat|moof|mdat|moof ...
+	  When a moof was received, it usually came with some extra bytes that would
+	  belong to the next mdat (because upstream doesn't care about atoms alignment).
+	  So those bytes were being discarded and playback would fail.
+	  This patch makes qtdemux store those extra bytes to reuse them later after the
+	  mdat is emptied.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=710623
+
+2013-11-07 09:49:55 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiudpsink: Also use the bind-port property if no bind-address was given
+
+2013-11-07 00:51:12 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>
+
+	* sys/osxaudio/gstosxcoreaudiohal.c:
+	  osxaudiosink: fix segfault when we can't get the channels layout
+
+2013-11-05 17:26:49 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpvp8pay.c:
+	  rtpvp8pay: Make Picture ID mode configurable and default to no picture ID
+	  Some implementations (linphone) only support no picture at all in the
+	  stream and will fail if one is provided.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=711497
+
+2013-11-05 11:18:34 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From 865aa20 to dbedaa0
+
+2013-01-29 10:51:07 +0100  Paul HENRYS <visechelle@gmail.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	  Add call to gst_rtp_h264_pay_clear_sps_pps() when receiving a STREAM_START event
+	  https://bugzilla.gnome.org/show_bug.cgi?id=692787
+
+2013-11-02 22:50:47 +0100  Rico Tzschichholz <ricotz@ubuntu.com>
+
+	* gst/rtsp/Makefile.am:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtsp: Add missing gio-2.0 deps and includes
+
+2013-11-01 18:31:36 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiofx/audioiirfilter.c:
+	  audioiirfilter: Fix initialization coefficient handling
+	  Broke unit test.
+
+2013-10-31 14:05:43 -0700  Aleix Conchillo Flaque <aleix@oblong.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: allow setting tls certificate validation flags
+	  Added a new property "tls-validation-flags". If the url transport is
+	  TLS, the validation flags will be set to the rtsp connection.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=711230
+
+2013-10-31 22:43:49 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiofx/audiofxbaseiirfilter.c:
+	* gst/audiofx/audioiirfilter.c:
+	  audioiirfilter: Don't crash if no filter coefficients are provided
+	  ...and by default use a identity filter.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=710215
+
+2013-10-31 19:15:12 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/wavpack/gstwavpackenc.c:
+	  wavpackenc: Fix writing of MD5 sums and other metadata blocks
+	  These don't have the FINAL_BLOCK flag set.
+
+2013-10-31 13:02:11 -0200  Djalma Lúcio Soares da Silva <dlucio@impa.br>
+
+	* ext/raw1394/gsthdv1394src.c:
+	  hdv1394src: Make it possible to select a camera by its GUID
+	  The source hdv1394src has the guid property that permits select a camera
+	  connected from its GUID number.
+	  However when this property is setted the selected camera is not changed.
+	  The source continues using the default camera.
+	  This problem was solved using the function iec61883_cmp_connect.
+	  The reference for the function could be found here:
+	  http://www.dennedy.org/libiec61883/API-iec61883-cmp-connect.html
+	  The solution came from dvgrab source code.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=710415
+
+2013-10-31 13:20:41 -0300  Thiago Santos <ts.santos@partner.samsung.com>
+
+	* tests/check/elements/souphttpsrc.c:
+	  tests: souphttpsrc: add explicit cast to silence warning
+	  Silencing this warning:
+	  elements/souphttpsrc.c:533:14: error: comparison between ‘SoupKnownStatusCode’ and ‘enum <anonymous>’ [-Werror=enum-compare]
+	  if (status != SOUP_STATUS_OK && !send_error_doc)
+	  With gcc 4.8.2 (debian)
+
+2013-10-31 10:38:35 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: proxy new buffer mode
+
+2013-10-30 16:49:36 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  jitterbuffer: add new timestamp mode
+	  Add a new timestamp mode that assumes the local and remote clock are
+	  synchronized. It takes the first timestamp as a base time and then uses the RTP
+	  timestamps for the output PTS.
+
+2013-10-30 22:12:45 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: Fix compiler warning
+	  matroska-demux.c: In function 'gst_matroska_demux_add_stream':
+	  matroska-demux.c:1379:7: error: format '%u' expects argument of type 'unsigned int', but argument 4 has type 'guint64' [-Werror=format=]
+	  "%03u", context->uid);
+	  ^
+
+2013-10-28 13:21:15 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* gst/videomixer/videoconvert.c:
+	  videomixer: remove unneeded guint comparaison
+	  https://bugzilla.gnome.org/show_bug.cgi?id=711010
+
+2013-10-28 14:13:12 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* sys/osxvideo/cocoawindow.h:
+	* sys/osxvideo/cocoawindow.m:
+	  osxvideosink: fix missing selector name warnings
+	  The spaces matter in ObjC
+	  https://bugzilla.gnome.org/show_bug.cgi?id=711013
+
+2013-10-28 13:31:34 +0000  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* gst/y4m/gsty4mencode.c:
+	  y4menc: fix uninitialized variable warning
+	  https://bugzilla.gnome.org/show_bug.cgi?id=711011
+
+2013-10-25 11:30:36 -0300  Thiago Santos <ts.santos@partner.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: check if the end_time is defined before using it
+	  Avoids sending EOS too soon because of overflow. Can happen on
+	  fragmented mp4 playback.
+
+2013-10-23 13:38:20 -0300  Thiago Santos <ts.santos@partner.samsung.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: use correct unref function
+	  Events aren't GstObjects, but GstMiniObjects
+
+2013-10-15 08:16:20 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: rename chunks_are_chunks to chunks_are_samples and flip the logic
+	  As the variable name suggests, sometimes chunks are chunks. Rename the variable
+	  to tell what they are when they are not chunks.
+
+2013-10-09 08:04:20 +0200  Stefan Sauer <ensonic@users.sf.net>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix typos and add more logging for unhandled parts
+
+2013-10-14 16:23:25 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiudpsink: Fix memory leak
+	  Unmap all GstMemory of the current buffer when flushing.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=710110
+
+2013-10-12 20:44:31 +0100  Tim-Philipp Müller <tim@centricular.net>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: fix broken sample pipeline
+	  which was muxing raw audio and video into flvmux, which won't work,
+	  even if there were converters.
+
+2013-10-12 20:37:41 +0100  Tim-Philipp Müller <tim@centricular.net>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: require stream-format=raw for mpeg-2 too, but don't require framed field
+	  raw implies that it's framed already. Fixes .. ! faac ! flvmux
+
+2013-10-07 14:27:21 -0300  Thiago Santos <ts.santos@partner.samsung.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: do not emit EOS when connection drops
+	  If the pipeline is stalled for too long, souphttpsrc will block and
+	  stop fetching data from the network. This can cause the connection to
+	  drop and souphttpsrc would handle it as an EOS. This patch makes it
+	  persist and try to fetch more data until the end of the content length
+	  or until receiving an error that it is beyong limits in case the content
+	  is unknown.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=683536
+
+2013-10-10 13:52:35 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* ext/dv/gstdvdec.c:
+	* ext/dv/gstdvdec.h:
+	  dvdec: Don't send segment event before caps
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709728
+
+2013-10-09 17:46:33 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: Send stream-start, caps and segment events in the right order
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709728
+
+2013-10-08 11:28:04 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* gst/wavenc/gstwavenc.c:
+	  wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709614
+
+2013-10-07 12:54:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* gst/deinterlace/tvtime/greedyh.c:
+	  deinterlace: Fix handling of planar video formats in greedyh method
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709507
+
+2013-10-06 10:01:26 -0700  Reynaldo H. Verdejo Pinochet <r.verdejo@partner.samsung.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska: Trivial grammar fix on debug msg
+
+2013-10-06 09:17:00 -0700  Reynaldo H. Verdejo Pinochet <r.verdejo@partner.samsung.com>
+
+	* gst/matroska/matroska-mux.c:
+	* gst/matroska/matroska-mux.h:
+	* gst/matroska/webm-mux.c:
+	  matroskamux: Add context flag for WebM
+	  WebM has a couple of specific requirements we need to handle.
+	  Idea is to set this flag once and just rely on mux->is_webm
+	  at run time instead of repeatedly figuring this out from
+	  GST_MATROSKA_DOCTYPE_WEBM (which requires a strcmp()).
+
+2013-10-04 14:42:59 -0700  Reynaldo H. Verdejo Pinochet <r.verdejo@partner.samsung.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska: Do not write SegmentUID for WebM mux
+	  WebM spec states SegmentUID is Unsupported. Files produced
+	  with gstreamer without this change will spit an error like
+	  this when passed to mkvalidator:
+	  ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192
+
+2013-10-05 00:00:03 +0200  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: make dvd palette change event sticky
+	  So they don't get lost.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709454
+
+2013-10-03 16:39:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/videofilter/gstvideoflip.c:
+	* gst/videofilter/gstvideoflip.h:
+	  videoflip: Add automatic flip mode driven by image-orientation tag
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709312
+
+2013-10-04 13:34:09 +0200  Peter Korsgaard <peter@korsgaard.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: O_CLOEXEC needs _GNU_SOURCE
+	  On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only
+	  defined when _GNU_SOURCE is specified, so do so.
+	  _GNU_SOURCE needs to be defined before any system headers are included,
+	  so move the fcntl.h section up.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709423
+
+2013-10-04 12:11:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: fix race in flush-start/flush-stop
+	  When flush-stop arrives before we process the result of the _push() in the
+	  loop function, we might pause even though we are not flushing anymore. Fix this
+	  race by waiting for the srcpad loop function to completely pause after doing the
+	  flush-start.
+
+2013-10-03 22:38:43 +0200  Mathieu Duponchelle <mduponchelle1@gmail.com>
+
+	* gst/videomixer/videoconvert.c:
+	  videomixer: Update videoconvert copy
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709390
+
+2013-10-03 21:36:34 +0200  Mathieu Duponchelle <mduponchelle1@gmail.com>
+
+	* gst/videomixer/videomixer2.c:
+	  videomixer: Check if the pad needs reconfiguration in collected
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709384
+
+2013-10-03 14:39:35 +0100  Matthieu Bouron <matthieu.bouron@collabora.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: Relax sink caps
+	  Since jpegdec already parse the jpeg stream, the sink caps could be
+	  relaxed. This will allow jpegdec to be selected in more case and in
+	  particular when the jpeg typefinder does not find the width and height.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709352
+
+2013-10-03 18:33:01 +0100  Tim-Philipp Müller <tim@centricular.net>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2src: print probed caps as caps again in debug log
+	  This got lost during refactoring.
+
+2013-10-03 11:59:25 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Add support for the mp2v fourcc for MPEG-2 video
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709270
+
+2013-10-02 15:56:53 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Fix memory leak
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709266
+
+2013-09-30 12:31:42 +0300  Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_fourcc.h:
+	* gst/isomp4/qtdemux_types.c:
+	  qtdemux: Add HEVC support
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709093
+
+2013-09-30 12:24:32 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/rtp/gstrtpgstpay.c:
+	  rtpgstpay: Fix memory leak
+	  We were leaking the GList nodes of the pending buffers.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=709079
+
+2013-09-30 12:31:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtpjitterbuffer.h:
+	  rtpjitterbuffer: fix race when updating the next_seqnum
+	  If we were not waiting for the missing seqnum when we insert the lost packet
+	  event in the jitterbuffer, we end up not updating the next_seqnum and wait
+	  forever for the lost packets to arrive. Instead, keep track of the amount of
+	  packets contained by the jitterbuffer item and update the next expected
+	  seqnum only after pushing the buffer/event. This makes sure we correctly handle
+	  GAPS in the sequence numbers.
+
+2013-09-30 12:30:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: small debug improvement
+
+2013-09-30 11:53:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: reset skew does not reset clock-rate
+	  Don't reset the clock-rate when we reset the skew correction algorithm.
+	  Reset the skew correction algorithm when we change the clock-rate.
+
+2013-09-30 11:16:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: pause timer when PAUSED
+	  Also pause the timer when we go to the PAUSED state. It is possible that we
+	  don't have a clock or base-time in PAUSED to perform the timeouts.
+
+2013-09-30 11:15:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: improve debug
+
+2013-09-26 20:41:26 +0200  Hans Månsson <hansm@axis.com>
+
+	* gst/isomp4/gstqtmuxmap.c:
+	  mp4mux: Do not require framerate in peer video caps
+	  Remove the framerate restriction on the caps.
+	  Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864
+
+2013-09-27 15:05:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: also go into the loop function after connect
+	  When we have opened the stream, go into the loop function so that we can
+	  receive messages from the server.
+
+2013-09-27 12:53:06 +0200  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8
+	  https://bugzilla.gnome.org/show_bug.cgi?id=707933
+
+2013-09-26 16:20:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: don't calculate skew without rtptime
+	  Skip trying to calculate the skew when we don't have an rtptime.
+	  It causes problems when lost packet events are placed in the jitterbuffer.
+
+2013-09-25 23:46:14 +0100  Tim-Philipp Müller <tim@centricular.net>
+
+	* configure.ac:
+	  configure: get rid of AS_SCRUB_INCLUDE
+	  Should not be needed any more.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=707658
+
+2013-09-25 17:42:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: disable checks when linking pads
+	  We know the pad links will work (and we don't check the return value
+	  anyway).
+
+2013-09-25 17:36:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: avoid some pad link checks
+	  Link pads without checks, we know it will work.
+
+2013-09-25 12:55:21 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Don't error out if downstream is not seekable for non-fragmented variants
+	  Doing so would be a regression over 1.0 and breaks the unit test.
+	  However the result will be most likely unusable, so let's post
+	  a warning message on the bus.
+
+2013-09-24 04:02:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: calculate some stats
+
+2013-09-23 17:05:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: move send_lost_event function
+	  Move the send_lost_event function to the do_lost_event handling, there is no
+	  need to have a separate function.
+
+2013-09-16 11:20:51 -0300  Thiago Santos <thiago.sousa.santos@collabora.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: add code to parse creation time earlier than 1970
+	  Use g_date_time seconds manipulation to allow to cover the quicktime
+	  spec for creation_time. It uses seconds since 1904.
+	  Both paths could be done using the generic approach of seconds since
+	  1904 with GDateTime handling, but the first path using seconds from
+	  1970 should be more commonly found and avoids a few objects creation and
+	  ref/unref, so keep it there for performance.
+	  Additionally, the code for handling seconds since 1970 changed from >
+	  to >= because having 0 seconds since 1970 is also a valid case for that
+	  path to handle.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=707975
+
+2013-09-21 00:55:26 +0200  Matej Knopp <matej.knopp@gmail.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily
+	  https://bugzilla.gnome.org/show_bug.cgi?id=708505
+
+2013-09-24 18:30:04 +0100  Tim-Philipp Müller <tim@centricular.net>
+
+	* README:
+	* common:
+	  Automatic update of common submodule
+	  From 6b03ba7 to 865aa20
+
+2013-09-24 15:05:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* configure.ac:
+	  configure: Actually use 1.3.0.1 as version to make configure happy
+
+2013-09-24 15:00:24 +0200  Sebastian Dröge <slomo@circular-chaos.org>
+
+	* configure.ac:
+	  Back to development
+
 === release 1.2.0 ===
 
-2013-09-24  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2013-09-24 14:21:08 +0200  Sebastian Dröge <slomo@circular-chaos.org>
 
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
 	* configure.ac:
-	  releasing 1.2.0
+	* 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:
+	* win32/common/config.h:
+	  Release 1.2.0
+
+2013-09-24 14:20:51 +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
 
 2013-09-20 19:43:21 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>
 
diff --git a/NEWS b/NEWS
index 56255fa9f..5fc86c82d 100644
--- a/NEWS
+++ b/NEWS
@@ -1,123 +1,110 @@
-This is GStreamer Good Plugins 1.2.0
+This is GStreamer Good Plugins 1.3.1
 
-Changes since 1.0:
+Changes since 1.2:
 
 New API:
- • GstContext negotiation / sharing / announcing for sharing a
-   generic context between elements, e.g. a display handle
- • GL texture upload conversion meta for allowing different
-   buffer types to be converted to an OpenGL texture
- • GstCapsFeatures as extension to GstCaps for allowing the
-   negotiation of specific memory or meta requirements between
-   elements
- • GstMemory flags for contiguous and non-mappable memory
- • The stream-start event has optional flags now, e.g. for signalling
-   sparse streams
- • The stream-start even has an optional group-id field now to signal
-   all streams that should be played together
- • Allocators library in gst-plugins-base, currently only with generic
-   dmabuf memory support
- • insertbin library for easier handling of dynamically linked
-   pipelines (in -bad for now)
- • EGL helper library (in -bad for now)
- • MPEG-TS data structure library (in -bad for now)
- • New GstVideoRegionOfInterestMeta to describe a region of interest on
-   video frames.
- • GstVideoDecoder/Encoder has new ::flush() vfunc to replace the
-   ill-defined ::reset() vfunc.
- • The URI query allows to query the redirected URI now.
+ • GstMessageType has GST_MESSAGE_EXTENDED added. All types before
+   that can be used together as a flags type as before, but from
+   that message onwards the types are just counted incrementally.
+   This was necessary to be able to add more message types.
+   In 2.0 GstMessageType will just become an enum and not a flags
+   type anymore.
+ • GstDeviceMonitor for device probing, e.g. to list all available
+   audio or video capture devices. This is the replacement for
+   GstPropertyProbe from 0.10.
+ • Events accumulate the running-time offset now when travelling
+   through pads, as set by the gst_pad_set_offset() function. This
+   allows to compensate for this in the QOS event for example.
+ • GstBuffer has a new flag "tag-memory" that is set automatically
+   when memory is added or removed to a buffer. This allows buffer
+   pools to detect if they can recycle a buffer or need to reset
+   it first.
+ • GstToc has new API to mark GstTocEntries as loops.
+ • A not-authorized resource error has been defined to notify
+   applications that accessing the resource has failed because
+   of missing authorization and to distinguish this case from others.
+   This change is actually already in 1.2.4.
+ • GstPad has a new flag "accept-intersect", that will let the default
+   ACCEPT_CAPS query handler do an intersection instead of subset check.
+   This is interesting for parser elements that can handle incomplete
+   caps.
+ • GstCollectPads has support for flushing and a default handler for
+   SEEK events now.
+ • GstSegment has new API to offset the running time by a specific
+   value and this is used in GstPad to allow positive and negative
+   offsets in gst_pad_set_offset() in all situations.
+ • Support for h265/HEVC and VP8 has been added to the codec utils and codec
+   parsers library, and was integrated into various elements.
+ • API for adjusting the TLS validation of RTSP connection has been added.
+ • The RTSP and SDP library has MIKEY (RFC 3830) support now, and
+   there is API to distinguish between the different RTSP profiles.
+ • API to access RTP time information and statistics.
+ • Support for auxiliary streams was added to rtpbin.
+ • Support for tiled, raw video formats has been added.
+ • GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
+   events and merge custom tags into them consistently.
+ • playbin/playsink has support for application provided audio and video
+   filters.
+ • The GL library was merged from gst-plugins-gl to gst-plugins-bad,
+   providing a generic infrastructure for handling GL inside GStreamer
+   pipelines and a plugin with some elements using these, especially
+   a video sink. Supported platforms currently are Android, Cocoa (OS X),
+   DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
+   Wayland and EGL platforms.
+   This replaces eglglessink and also is supposed to replace osxvideosink.
 
-Major changes:
- • New tool: gst-play-1.0 in gst-plugins-base for basic playback
-   testing on the command line.
- • New plugins:
-   ∘ mssdemux for Microsoft Smooth Streaming
-   ∘ dashdemux for DASH adaptive streaming protocol
-   ∘ bluez for interaction with Bluetooth devices
-   ∘ openjpeg for JPEG2000 decoding and encoding
-   ∘ daala for experimental Daala decoding and encoding
-   ∘ vpx plugin has experimental VP9 decoding and encoding support
-   ∘ webp plugin for WebP decoding (encoding to be added later)
-   ∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse,
-     mfc, ivtv, accuraterip and audiofxbad
-
- • Moved plugins:
-   ∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in
-     gst-plugins-good now
-
- • Video:
-   ∘ Fix handling of interlaced video in converters such as videoscale
-     and videoconvert (e.g. scale both fields independently)
-   ∘ videoconvert will try harder to minimise quality losses when
-     conversion is necessary
-   ∘ The experimental GstSurfaceConverter, GstSurfaceMeta and
-     GstVideoContext APIs from the (confusingly-named) 
-     libgstbasevideo-1.0 library in gst-plugins-bad have now been
-     removed and been replaced by new APIs in GStreamer Core and
-     gst-plugins-base (see above). Since that was all that was left in
-     this library, the entire experimental libgstbasevideo-1.0 library
-     has been removed from gst-plugins-bad
-   ∘ Chroma subsampling and chroma siting conversion is better handled
-     in videoconvert and the support for interlaced video was improved.
-   ∘ New pinwheel and spoke patterns in videotestsrc
-   ∘ videomixer can now accept different video formats on its sinkpads
-     and converts to a common format during mixing
 
- • Audio:
-   ∘ audioconvert will try harder to minimise quality losses when
-     conversion is necessary
-   ∘ adder now allows muting/unmuting of its input streams, and also
-     per-input stream volume 
-   ∘ pulseaudio elements can switch between devices during playback now
-   ∘ aacparse can convert between ADTS←→RAW
-
- • Platform specific changes:
-   ∘ Caps, events, etc. are now printed in the GStreamer debug logs
-     with their content instead of just the pointer address even on
-     non-glibc platforms (e.g. Windows, OSX, Android).
-   ∘ Network elements (UDP/TCP) now work better with platforms,
-     where IPv6 sockets can't handle IPv4 (e.g. Windows)
-   ∘ Linux/BSD: v4l2 had many improvements and cleanups
+Major changes:
+ • New plugins and elements:
+   ∘ v4l2videodec element for accessing hardware codecs on
+     platforms that make them accessible via V4L2, e.g.
+     Samsung Exynos. This comes together with major refactoring
+     of the existing V4L2 elements and the corresponding
+     infrastructure.
+     The v4l2videodec element replaces the mfcdec element.
+   ∘ rtpstreampay and rtpstreamdepay elements for transmitting
+     RTP packets over a stream API (e.g. TCP) according to
+     RFC 4571.
+   ∘ rtprtx elements for standard compliant implementation of
+     retransmissions, integrated into the rtpmanager plugin.
+   ∘ audiomixer element that mixes multiple audio streams together
+     into a single one while keeping synchronization. This is
+     planned to become the replacement of the adder element.
+   ∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
+   ∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
+   ∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
+   ∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
+   ∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
+     are available on OS X and iOS now.
 
  • Other changes:
-   ∘ gst-libav now uses libav 9
-   ∘ Static linking of plugins is supported now (also in 1.0.7)
-   ∘ rtspsrc: add support for NetClientClock: when the server suggests a
-     GstNetTimeProvider in the SDP, set up a GstNetClientClock that
-     slaves to the remote clock and suggest this clock in provide_clock.
-     Simplifies synchronized playback of a resource from an RTSP server.
-     gst-rtsp-server now supports adding this to the SDP and can provide
-     a network clock
-   ∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements
-   ∘ SRTP and DTLS support
-   ∘ Changes to many elements and core to use the correct sticky event
-     order and also not lose any important sticky events during flushing
-   ∘ >1000 fixed bug reports, and many other bug fixes and other
-     improvements everywhere that had no bug report
+   ∘ gst-libav now uses libav 10, and gained support for H265/HEVC.
+   ∘ Support for hardware codecs and special memory types has been
+     improved with bugfixes and feature additions in various plugins
+     and base classes.
+   ∘ Various bugfixes and improvements to buffering in queue2 and
+     multiqueue elements.
+   ∘ dvbsrc supports more delivery mechanisms and other features
+     now, including DVB S2 and T2 support.
+   ∘ The MPEGTS library has support for many more descriptors.
+   ∘ Major improvements to tsdemux, especially time related.
+   ∘ souphttpsrc now has support for keep-alive connections,
+     compression, configurable number of retries and configuration
+     for SSL certificate validation.
+   ∘ hlsdemux has undergone major refactoring and works more
+     reliable now and supports more HLS features like trick modes.
+     Also fragments are pushed downstream while they're downloaded
+     now instead of waiting for each fragment to finish.
+   ∘ videoflip can automatically flip based on the orientation tag.
+   ∘ openjpeg supports the OpenJPEG2 API.
+   ∘ gst-rtsp-server supports SRTP and MIKEY now.
+   ∘ Lots of fixes for coverity warnings all over the place.
+   ∘ 400+ fixed bug reports, and many other bug fixes and other
+     improvements everywhere that had no bug report.
 
 Things to look out for:
- • Single header includes for all libraries, e.g. #include
-   <gst/video/video.h> - this was needed for some bindings.
- • Stricter (correct) caps subset checking in some cases where this was
-   not correct before. Caps will now always fail to be a compatible
-   subset of another set of caps if the subset caps are missing some
-   fields that the superset caps have. This might lead to not-negotiated
-   errors if caps are incomplete now. However, it also prevents possible
-   data corruption caused by piping data formatted in an
-   incompatible/unexpected way into some elements. Check your h264 caps
-   for stream-format and alignment fields and AAC caps for the
-   stream-format field. This change will also be included in the next
-   stable 1.0.8 release.
- • Stricter checking for missing events and correct sticky event order
-   (stream-start, caps, segment) in some places; this is not enabled in
-   stable releases by default, but you may get warnings when using git
-   builds, development releases or when compiling with
-   -UG_DISABLE_ASSERT in CFLAGS
- • x264enc now outputs data in byte-stream by default if downstream has
-   ANY caps (e.g. appsink without caps set, filesink, udpsink,
-   tcpserversink etc.)
- • The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a
-   different format now. This new format uses the data structures from
-   the new MPEGTS library
- • The GstContext API has changed between 1.1.4 and 1.1.90
+ • The eglglessink element was removed and replaced by the glimagesink
+   element.
+ • The mfcdec element was removed and replaced by v4l2videodec.
+ • osxvideosink is only available in OS X 10.6 or newer.
 
diff --git a/RELEASE b/RELEASE
index 3a80727ac..30d759566 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,16 +1,29 @@
 
-Release notes for GStreamer Good Plugins 1.2.0
+Release notes for GStreamer Good Plugins 1.3.1
 
 
-The GStreamer team is proud to announce a new feature release
-in the 1.x stable series of the
-core of the GStreamer streaming media framework.
+The GStreamer team is pleased to announce the first release of the unstable
+1.3 release series. The 1.3 release series is adding new features on top of
+the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
+series of the GStreamer multimedia framework. The unstable 1.3 release series
+will lead to the stable 1.4 release series in the next weeks, and newly added
+API can still change until that point.
 
 
-The 1.x series is a stable series targeted at end users.
-It is not API or ABI compatible with the stable 0.10.x series.
-It is, however, parallel installable with the 0.10.x series and
-will not affect an existing 0.10.x installation.
+
+Binaries for Android, iOS, Mac OS X and Windows will be provided separately
+during the unstable 1.3 release series.
+
+
+
+The versioning scheme that is used in general is that 1.x.y is API and
+ABI backwards compatible with previous 1.x.y releases. If x is an even
+number it is a stable release series and all releases in this series
+will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
+x is odd it is a development release series that will lead to the next
+stable release series 1.x+1 and contains new features and bigger
+changes. During the development release series, new API can still
+change.
 
 
 
@@ -57,11 +70,101 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
 
 Bugs fixed in this release
      
-      * 706083 : v4l2src: UVC Allocated buffers wrapped in GstBuffer get orphaned by GstBuffer API
-      * 707242 : qtmux: streamable and faststart property have no effect
-      * 707933 : matroskademux: Wrong UTF8 detection causes wrong detection of subtitle encoding
-      * 708501 : osxvideosink: fix segfault releasing the element
-      * 708622 : rtpjitterbuffer: fix various regressions
+      * 728501 : rtpaux/rtprtx: Unit tests are racy and take very long sometimes
+      * 719636 : deinterlace: alters caps in passthrough mode preventing hardware decode
+      * 727305 : matroskademux: Add support for A_OPUS
+      * 725632 : v4l2: Normalise control names in the same way as v4l2-ctl
+      * 345830 : qtdemux: better edit lists handling
+      * 636143 : avidemux: report creation date/time via GST_TAG_DATE_TIME
+      * 652986 : rtpjitterbuffer: events are not serialized
+      * 664339 : matroskamux: support for audio/x-adpcm
+      * 691570 : [isomp4/qtdemux] lots of critical warnings on this sample file
+      * 692787 : rtph264pay: No way to clear SPS and PPS in case of a new stream
+      * 705024 : aacparse: does not propagate downstream sample rate restriction upstream
+      * 705982 : mp4mux: HDLR box name string is not NULL terminated
+      * 708165 : videomixer: Store and forward tag events
+      * 709079 : rtpgstpay: Leaks memory
+      * 709093 : qtdemux: add HEVC support
+      * 709266 : matroska-demux leaks memory
+      * 709312 : videoflip: Add an automatic method that flip base on image-orientation tag
+      * 710415 : hdv1394src: Not possible to select a HDV camera from GUID
+      * 710762 : qtdemux: fails reading some MOV files with problematic jpeg frames
+      * 711010 : videomixer: remove unneeded guint comparison
+      * 711011 : y4mencode: fix uninitialized variable warning
+      * 711013 : osxvideosink: fix missing selector name warning
+      * 711084 : rtpmanager: add new rtprtxsend and rtprtxreceive elements for retransmission
+      * 711087 : rtpbin: Support Auxiliary streams
+      * 711270 : check: add rtpsession test
+      * 711411 : rtpjitterbuffer: implement RTX statistics
+      * 711412 : rtpjitterbuffer: Automatically calculate RTX properties based on RTT
+      * 711560 : rtpsession: ssrc collision improvements
+      * 711693 : rtpsession: Implement various session statistics
+      * 712206 : v4l2: print FOURCC before enumerating
+      * 712254 : multifilesrc: Implement seeking in case of multiple images
+      * 712303 : qtdemux: playback regression after commit ae1150e85cf99d7482933aa6f7e4f012fe45a3ec
+      * 712567 : rtpsession: RBs are not included in SRs after the first RTCP timeout when there are multiple internal senders
+      * 712612 : v4l2bufferpool: take over the floating reference for the new allocator
+      * 712754 : v4l2: add support for multi-planar V4L2 API
+      * 719434 : rtph264pay maps and unmaps inbuffer twice
+      * 719497 : videoflip: crashes on tag list without orientation tag in git master
+      * 719783 : qtdemux: regression with mp4-main-multi-mpd-AV-NBS.mpd
+      * 719829 : rtp: Add RFC4571 framing/de-framing element
+      * 719938 : rtpbin: allow dynamic RTP/RTCP encoders and decoders
+      * 720371 : rtpbin: Impossible to set jitterbuffer rtx-* properties when using rtpbin
+      * 720512 : flacparse: Doesn't set the codec tag
+      * 720568 : v4l2: Various changes to allow using M2M decoders
+      * 720995 : matroskamux: add g726 adpcm support
+      * 721245 : osxvideosink: Fails to build on OS X Leopard 10.5.8
+      * 721342 : shout2send: Some minor cleanups
+      * 722175 : rtpmanager: improve code of rtprtx* elements
+      * 722370 : rtprtxsend: push rtx buffers from a different thread to avoid long retransmission delays
+      * 722372 : rtpjitterbuffer: Got data flow before stream-start event
+      * 722394 : v4l2: set GST_BUFFER_FLAG_DELTA_UNIT when appropriate
+      * 722396 : avimux: don't make the buffer writable unless absolutely necessary
+      * 722866 : rtspsrc: add rtpjitterbuffer do-retransmission property
+      * 722981 : autodetect elements have inconsistent handling of autoplugin
+      * 723166 : qtdemux: incorrect tag character
+      * 723269 : matroskamux: used uid list grows forever
+      * 723289 : cairooverlay: add RGB16 support
+      * 723502 : gst-plugins-good: Do not build check tests for disabled plugins
+      * 723849 : matroska: add support for GRAY8, BGR and RGB video  colourspaces in V_UNCOMPRESSED codec
+      * 724085 : gst-plugins-good/docs: Rebalance docbook < para > tags in comments
+      * 724213 : rtph264pay: shouldn't update time for sending SPS and PPS if we failed to send SPS or PPS
+      * 724396 : rtspsrc: add tls-database property
+      * 724636 : v4l2videodec: VP8 KO with playbin
+      * 724705 : videomixer: Port to new collectpads API
+      * 724712 : rtspsrc doesn't set caps on pads before adding them
+      * 724899 : v4l2src does not set interlaced flag on buffers
+      * 725008 : matroskademux: crash with 24bit raw audio
+      * 725159 : rtpjitterbuffer: RTP sequence number rollover problems
+      * 725361 : [regression] rtpsession: setting the " internal-ssrc " property does nothing
+      * 725480 : gst-plugins-good: Ignore gcov intermediate files
+      * 725723 : osxvideo: GetCurrentProcess not available on Mavericks
+      * 725948 : videomixer2: crash after renegotiating with different resolution
+      * 726106 : matroskademux: does not handle fps lower than 1
+      * 726161 : png plugin handles interlaced png files incorrectly
+      * 726696 : rtspsrc memleaks
+      * 726737 : osxvideosink: advertize for the video meta API support
+      * 726738 : osxvideosink: use the video frame API instead of the video meta API
+      * 726833 : ximagesrc: Add alpha channel support
+      * 726837 : rtspsrc segfault
+      * 727821 : souphttpsrc: Regression in push mode
+      * 727867 : qtdemux: Does not return stream flags from trex atom
+      * 727878 : qtdemux: replace duplicated variable when parsing trex atom
+      * 729223 : wavparse: drops upstream tags for .wav files that are ID3 tagged
+      * 728987 : qtdemux: 'caps' may be used uninitialized in this function.
+      * 729067 : goom filter: left shift of 24 places cannot be represented in type 'int'
+      * 712333 : regression: videoflip: aborts with gst_video_flip_transform_caps: code should not be reached
+      * 722077 : v4l2: compile error - 'V4L2_CAP_VIDEO_M2M_MPLANE' undeclared
+      * 722127 : v4l2: Add NV12_64Z32 support
+      * 722128 : v4l2: Implement video decoder
+      * 723446 : v4l2src: Should detect support for mplanar formats during runtime
+      * 726453 : v4l2 plugin broken due to bundeling of videodev2.h
+      * 721764 : souphttpsrc: Add ability to do HTTP session logging
+      * 722311 : matroskaparse: should try to identify data on stream header before going with a blind
+      * 722705 : Factor out common init/reset code from matroska parse/demux
+      * 712643 : qtdemux: couple of issues with vobsub
+      * 581295 : mp4mux:  Add support for embedded subtitles
 
 ==== Download ====
 
@@ -98,11 +201,70 @@ subscribe to the gstreamer-devel list.
         
 Contributors to this release
     
+      * Akihiro Tsukada
+      * Aleix Conchillo Flaque
+      * Aleix Conchillo Flaqué
+      * Alessandro Decina
+      * Alexander Zallesov
       * Andoni Morales Alastruey
+      * Benjamin Gaignard
+      * Branislav Katreniak
+      * Brendan Long
+      * Christian Fredrik Kalager Schaller
+      * Dan Kegel
+      * Darryl Gamroth
+      * Djalma Lúcio Soares da Silva
       * Edward Hervey
+      * George Kiagiadakis
+      * Göran Jönsson
+      * Hans Månsson
+      * Hugues Fruchet
+      * Jake Foytik
+      * Jan Schmidt
+      * Jeremy Huddleston
+      * Jeremy Huddleston Sequoia
+      * Jimmy Ohn
+      * Jonas Holmberg
+      * Josep Torra
+      * Julien Isorce
+      * Justin Joy
+      * Luis de Bethencourt
+      * Marc Leeman
+      * Mark Nauwelaerts
+      * Matej Knopp
+      * Mathieu Duponchelle
+      * MathieuDuponchelle
+      * Matthieu Bouron
+      * Michael Olbrich
+      * Mike Sheldon
+      * Nicola Murino
+      * Nicolas Dufresne
+      * Ognyan Tonchev
       * Olivier Crête
-      * Robert Krakora
+      * Paul HENRYS
+      * Per x Johansson
+      * Peter Korsgaard
+      * Philippe Normand
+      * Rafał Mużyło
+      * Reynaldo H. Verdejo Pinochet
+      * Rico Tzschichholz
+      * Ryan Lortie
+      * Santiago Carot-Nemesio
       * Sebastian Dröge
+      * Sebastian Rasmussen
+      * Simon Farnsworth
+      * Sreerenj Balachandran
+      * Stefan Sauer
+      * Stéphane Cerveau
+      * Thiago Santos
+      * Thibault Saunier
+      * Thijs Vermeir
       * Tim-Philipp Müller
+      * Todd Agulnick
+      * Torrie Fischer
+      * Vincent Penquerc'h
+      * William Jon McCann
+      * William Manley
       * Wim Taymans
+      * divhaere
  
\ No newline at end of file
diff --git a/configure.ac b/configure.ac
index 7b754d104..b31be2abc 100644
--- a/configure.ac
+++ b/configure.ac
@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
 dnl initialize autoconf
 dnl releases only do -Wall, git and prerelease does -Werror too
 dnl use a three digit version number for releases, and four for git/pre
-AC_INIT([GStreamer Good Plug-ins],[1.3.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
+AC_INIT([GStreamer Good Plug-ins],[1.3.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
 
 AG_GST_INIT
 
@@ -43,11 +43,11 @@ AC_DEFINE_UNQUOTED(GST_API_VERSION, "$GST_API_VERSION",
   [GStreamer API Version])
 
 AG_GST_LIBTOOL_PREPARE
-AS_LIBTOOL(GST, 300, 0, 300)
+AS_LIBTOOL(GST, 301, 0, 301)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.3.0.1
-GSTPB_REQ=1.3.0.1
+GST_REQ=1.3.1
+GSTPB_REQ=1.3.1
 
 dnl *** autotools stuff ****
 
diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args
index 6bdc4a575..8e17826f7 100644
--- a/docs/plugins/gst-plugins-good-plugins.args
+++ b/docs/plugins/gst-plugins-good-plugins.args
@@ -989,6 +989,26 @@
 </ARG>
 
 <ARG>
+<NAME>GstRTSPSrc::tls-database</NAME>
+<TYPE>GTlsDatabase*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>TLS database</NICK>
+<BLURB>TLS database with anchor certificate authorities used to validate the server certificate.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::tls-validation-flags</NAME>
+<TYPE>GTlsCertificateFlags</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>TLS validation flags</NICK>
+<BLURB>TLS certificate validation flags used to validate the server certificate.</BLURB>
+<DEFAULT>G_TLS_CERTIFICATE_UNKNOWN_CA|G_TLS_CERTIFICATE_BAD_IDENTITY|G_TLS_CERTIFICATE_NOT_ACTIVATED|G_TLS_CERTIFICATE_EXPIRED|G_TLS_CERTIFICATE_REVOKED|G_TLS_CERTIFICATE_INSECURE|G_TLS_CERTIFICATE_GENERIC_ERROR</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstRTPDec::skip</NAME>
 <TYPE>gint</TYPE>
 <RANGE></RANGE>
@@ -1264,7 +1284,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>ip</NICK>
-<BLURB>ip.</BLURB>
+<BLURB>IP address or hostname.</BLURB>
 <DEFAULT>"127.0.0.1"</DEFAULT>
 </ARG>
 
@@ -1334,7 +1354,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>url</NICK>
-<BLURB>url.</BLURB>
+<BLURB>the stream's homepage URL.</BLURB>
 <DEFAULT>""</DEFAULT>
 </ARG>
 
@@ -2432,7 +2452,7 @@
 <NAME>GstVideoFlip::method</NAME>
 <TYPE>GstVideoFlipMethod</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
+<FLAGS>rwx</FLAGS>
 <NICK>method</NICK>
 <BLURB>method.</BLURB>
 <DEFAULT>Identity (no rotation)</DEFAULT>
@@ -3214,7 +3234,7 @@
 <RANGE>>= G_MAXULONG</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Maximum Consecutive Decoding Errors</NICK>
-<BLURB>Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error).</BLURB>
+<BLURB>(Deprecated) Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -4165,7 +4185,77 @@
 <FLAGS>rw</FLAGS>
 <NICK>timeout</NICK>
 <BLURB>Value in seconds to timeout a blocking I/O (0 = No timeout).</BLURB>
-<DEFAULT>0</DEFAULT>
+<DEFAULT>15</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHTTPSrc::compress</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Compress</NICK>
+<BLURB>Allow compressed content encodings.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHTTPSrc::http-log-level</NAME>
+<TYPE>SoupLoggerLogLevel</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>HTTP log level</NICK>
+<BLURB>Set log level for soup's HTTP session log.</BLURB>
+<DEFAULT>SOUP_LOGGER_LOG_HEADERS</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHTTPSrc::keep-alive</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>keep-alive</NICK>
+<BLURB>Use HTTP persistent connections.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHTTPSrc::retries</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Retries</NICK>
+<BLURB>Maximum number of retries until giving up (-1=infinite).</BLURB>
+<DEFAULT>3</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHTTPSrc::ssl-ca-file</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>SSL CA File</NICK>
+<BLURB>Location of a SSL anchor CA file to use.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHTTPSrc::ssl-strict</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>SSL Strict</NICK>
+<BLURB>Strict SSL certificate checking.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHTTPSrc::ssl-use-system-ca-file</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Use System CA File</NICK>
+<BLURB>Use system CA file.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21335,7 +21425,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>RTX Delay</NICK>
 <BLURB>Extra time in ms to wait before sending retransmission event (-1 automatic).</BLURB>
-<DEFAULT>20</DEFAULT>
+<DEFAULT>-1</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21355,7 +21445,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>RTX Retry Period</NICK>
 <BLURB>Try to get a retransmission for this many ms (-1 automatic).</BLURB>
-<DEFAULT>160</DEFAULT>
+<DEFAULT>-1</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21365,7 +21455,17 @@
 <FLAGS>rw</FLAGS>
 <NICK>RTX Retry Timeout</NICK>
 <BLURB>Retry sending a transmission event after this timeout in ms (-1 automatic).</BLURB>
-<DEFAULT>40</DEFAULT>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::stats</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Statistics</NICK>
+<BLURB>Various statistics.</BLURB>
+<DEFAULT></DEFAULT>
 </ARG>
 
 <ARG>
@@ -21489,6 +21589,16 @@
 </ARG>
 
 <ARG>
+<NAME>GstRtpSession::stats</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Statistics</NICK>
+<BLURB>Various statistics.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstRtpRtxSend::rtx-payload-type</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
@@ -21503,7 +21613,7 @@
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Max Size Times</NICK>
+<NICK>Max Size Time</NICK>
 <BLURB>Amount of ms to queue (0 = unlimited).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
@@ -21511,11 +21621,11 @@
 <ARG>
 <NAME>GstRtpRtxSend::max-size-packets</NAME>
 <TYPE>guint</TYPE>
-<RANGE></RANGE>
+<RANGE><= 32767</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Max Size Packets</NICK>
 <BLURB>Amount of packets to queue (0 = unlimited).</BLURB>
-<DEFAULT>0</DEFAULT>
+<DEFAULT>100</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21533,12 +21643,32 @@
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>r</FLAGS>
-<NICK>Num RTX Packet</NICK>
-<BLURB>Number of retransmission packets sent.</BLURB>
+<NICK>Num RTX Packets</NICK>
+<BLURB> Number of retransmission packets sent.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
+<NAME>GstRtpRtxSend::payload-type-map</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Payload Type Map</NICK>
+<BLURB>Map of original payload types to their retransmission payload types.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpRtxSend::ssrc-map</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>w</FLAGS>
+<NICK>SSRC Map</NICK>
+<BLURB>Map of SSRCs to their retransmission SSRCs for SSRC-multiplexed mode (default = random).</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstRtpRtxReceive::rtx-payload-types</NAME>
 <TYPE>string</TYPE>
 <RANGE></RANGE>
@@ -21563,8 +21693,8 @@
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>r</FLAGS>
-<NICK>Num RTX Packet</NICK>
-<BLURB>Number of retransmission packets received.</BLURB>
+<NICK>Num RTX Packets</NICK>
+<BLURB> Number of retransmission packets received.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -21574,11 +21704,21 @@
 <RANGE></RANGE>
 <FLAGS>r</FLAGS>
 <NICK>Num RTX Associated Packets</NICK>
-<BLURB>correctly associated with retransmission requests.</BLURB>
+<BLURB>Number of retransmission packets correctly associated with retransmission requests.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
+<NAME>GstRtpRtxReceive::payload-type-map</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Payload Type Map</NICK>
+<BLURB>Map of original payload types to their retransmission payload types.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstV4l2Sink::device</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -23089,6 +23229,16 @@
 </ARG>
 
 <ARG>
+<NAME>GstSoupHttpClientSink::http-log-level</NAME>
+<TYPE>SoupLoggerLogLevel</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>HTTP log level</NICK>
+<BLURB>Set log level for soup's HTTP session log.</BLURB>
+<DEFAULT>SOUP_LOGGER_LOG_NONE</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstWavParse::ignore-length</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -24078,3 +24228,503 @@
 <DEFAULT>0</DEFAULT>
 </ARG>
 
+<ARG>
+<NAME>GstRtpVP8Pay::picture-id-mode</NAME>
+<TYPE>GstVP8RTPPayMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Picture ID Mode</NICK>
+<BLURB>The picture ID mode for payloading.</BLURB>
+<DEFAULT>No Picture ID</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::arnr-maxframes</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,15]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>AltRef max frames</NICK>
+<BLURB>AltRef maximum number of frames.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::arnr-strength</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,6]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>AltRef strength</NICK>
+<BLURB>AltRef strength.</BLURB>
+<DEFAULT>3</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::arnr-type</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[1,3]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>AltRef type</NICK>
+<BLURB>AltRef type.</BLURB>
+<DEFAULT>3</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::auto-alt-ref</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Auto alt reference frames</NICK>
+<BLURB>Automatically generate AltRef frames.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::buffer-initial-size</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Buffer initial size</NICK>
+<BLURB>Initial client buffer size (ms).</BLURB>
+<DEFAULT>4000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::buffer-optimal-size</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Buffer optimal size</NICK>
+<BLURB>Optimal client buffer size (ms).</BLURB>
+<DEFAULT>5000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::buffer-size</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Buffer size</NICK>
+<BLURB>Client buffer size (ms).</BLURB>
+<DEFAULT>6000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::cpu-used</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[-16,16]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>CPU used</NICK>
+<BLURB>CPU used.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::cq-level</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,63]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Constrained quality level</NICK>
+<BLURB>Constrained quality level.</BLURB>
+<DEFAULT>10</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::deadline</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Deadline</NICK>
+<BLURB>Deadline per frame (usec, 0=disabled).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::dropframe-threshold</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,100]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Drop Frame Threshold</NICK>
+<BLURB>Temporal resampling threshold (buf %).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::end-usage</NAME>
+<TYPE>GstVP9EncEndUsage</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Rate control mode</NICK>
+<BLURB>Rate control mode.</BLURB>
+<DEFAULT>Variable Bit Rate (VBR) mode</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::error-resilient</NAME>
+<TYPE>GstVP9EncErFlags</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Error resilient</NICK>
+<BLURB>Error resilience flags.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::horizontal-scaling-mode</NAME>
+<TYPE>GstVP9EncScalingMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Horizontal scaling mode</NICK>
+<BLURB>Horizontal scaling mode.</BLURB>
+<DEFAULT>Normal</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::keyframe-max-dist</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Keyframe max distance</NICK>
+<BLURB>Maximum distance between keyframes (number of frames).</BLURB>
+<DEFAULT>128</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::keyframe-mode</NAME>
+<TYPE>GstVP9EncKfMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Keyframe Mode</NICK>
+<BLURB>Keyframe placement.</BLURB>
+<DEFAULT>Determine optimal placement automatically</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::lag-in-frames</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,25]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Lag in frames</NICK>
+<BLURB>Maximum number of frames to lag.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::max-intra-bitrate</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max Intra bitrate</NICK>
+<BLURB>Maximum Intra frame bitrate.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::max-quantizer</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,63]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Maximum Quantizer</NICK>
+<BLURB>Maximum Quantizer (worst).</BLURB>
+<DEFAULT>63</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::min-quantizer</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,63]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Minimum Quantizer</NICK>
+<BLURB>Minimum Quantizer (best).</BLURB>
+<DEFAULT>4</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::multipass-cache-file</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Multipass Cache File</NICK>
+<BLURB>Multipass cache file.</BLURB>
+<DEFAULT>"multipass.cache"</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::multipass-mode</NAME>
+<TYPE>GstVP9EncMultipassMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Multipass Mode</NICK>
+<BLURB>Multipass encode mode.</BLURB>
+<DEFAULT>One pass encoding (default)</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::noise-sensitivity</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,6]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Noise sensitivity</NICK>
+<BLURB>Noise sensisivity (frames to blur).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::overshoot</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,1000]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Overshoot PCT</NICK>
+<BLURB>Datarate overshoot (max) target (%).</BLURB>
+<DEFAULT>100</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::resize-allowed</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Resize Allowed</NICK>
+<BLURB>Allow spatial resampling.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::resize-down-threshold</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,100]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Resize Down Threshold</NICK>
+<BLURB>Downscale threshold (buf %).</BLURB>
+<DEFAULT>60</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::resize-up-threshold</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,100]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Resize Up Threshold</NICK>
+<BLURB>Upscale threshold (buf %).</BLURB>
+<DEFAULT>30</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::sharpness</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,7]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sharpness</NICK>
+<BLURB>Filter sharpness.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::static-threshold</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Static Threshold</NICK>
+<BLURB>Motion detection threshold.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::target-bitrate</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Target bitrate</NICK>
+<BLURB>Target bitrate (in bits/sec).</BLURB>
+<DEFAULT>256000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::temporal-scalability-layer-id</NAME>
+<TYPE>GValueArray*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Coding layer identification</NICK>
+<BLURB>Sequence defining coding layer membership.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::temporal-scalability-number-layers</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[1,5]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Number of coding layers</NICK>
+<BLURB>Number of coding layers to use.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::temporal-scalability-periodicity</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,16]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Coding layer periodicity</NICK>
+<BLURB>Length of sequence that defines layer membership periodicity.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::temporal-scalability-rate-decimator</NAME>
+<TYPE>GValueArray*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Coding layer rate decimator</NICK>
+<BLURB>Rate decimation factors for each layer.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::temporal-scalability-target-bitrate</NAME>
+<TYPE>GValueArray*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Coding layer target bitrates</NICK>
+<BLURB>Target bitrates for coding layers (one per layer, decreasing).</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::threads</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,64]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Threads</NICK>
+<BLURB>Number of threads to use.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::timebase</NAME>
+<TYPE>GstFraction</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Shortest interframe time</NICK>
+<BLURB>Fraction of one second that is the shortest interframe time - normally left as zero which will default to the framerate.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::token-partitions</NAME>
+<TYPE>GstVP9EncTokenPartitions</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Token partitions</NICK>
+<BLURB>Number of token partitions.</BLURB>
+<DEFAULT>One token partition</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::tuning</NAME>
+<TYPE>GstVP9EncTuning</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Tuning</NICK>
+<BLURB>Tuning.</BLURB>
+<DEFAULT>Tune for PSNR</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::twopass-vbr-bias</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,100]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>2-pass VBR bias</NICK>
+<BLURB>CBR/VBR bias (0=CBR, 100=VBR).</BLURB>
+<DEFAULT>50</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::twopass-vbr-maxsection</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>2-pass GOP max bitrate</NICK>
+<BLURB>GOP maximum bitrate (% target).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::twopass-vbr-minsection</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>2-pass GOP min bitrate</NICK>
+<BLURB>GOP minimum bitrate (% target).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::undershoot</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,1000]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Undershoot PCT</NICK>
+<BLURB>Datarate undershoot (min) target (%).</BLURB>
+<DEFAULT>100</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Enc::vertical-scaling-mode</NAME>
+<TYPE>GstVP9EncScalingMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Vertical scaling mode</NICK>
+<BLURB>Vertical scaling mode.</BLURB>
+<DEFAULT>Normal</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Dec::deblocking-level</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 16</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Deblocking Level</NICK>
+<BLURB>Deblocking level.</BLURB>
+<DEFAULT>4</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Dec::noise-level</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 16</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Noise Level</NICK>
+<BLURB>Noise level.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Dec::post-processing</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Post Processing</NICK>
+<BLURB>Enable post processing.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Dec::post-processing-flags</NAME>
+<TYPE>GstVP9DecPostProcessingFlags</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Post Processing Flags</NICK>
+<BLURB>Flags to control post processing.</BLURB>
+<DEFAULT>Deblock|Demacroblock</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstVP9Dec::threads</NAME>
+<TYPE>guint</TYPE>
+<RANGE>[1,16]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max Threads</NICK>
+<BLURB>Maximum number of decoding threads.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
diff --git a/docs/plugins/gst-plugins-good-plugins.hierarchy b/docs/plugins/gst-plugins-good-plugins.hierarchy
index eecaeaeb7..dfe885968 100644
--- a/docs/plugins/gst-plugins-good-plugins.hierarchy
+++ b/docs/plugins/gst-plugins-good-plugins.hierarchy
@@ -14,6 +14,9 @@ GObject
           GstAudioClock
       GstControlBinding
       GstControlSource
+      GstDeviceMonitor
+        GstPulseDeviceMonitor
+        GstV4l2DeviceMonitor
       GstElement
         Gst3GPPMux
         GstALawDec
@@ -40,6 +43,7 @@ GObject
           GstDcaParse
           GstFlacParse
           GstMpegAudioParse
+          GstRtpStreamDepay
           GstSbcParse
           GstWavpackParse
         GstBaseSink
@@ -137,10 +141,11 @@ GObject
             GstWarpTV
         GstBin
           GstAspectRatioCrop
-          GstAutoAudioSink
-          GstAutoAudioSrc
-          GstAutoVideoSink
-          GstAutoVideoSrc
+          GstAutoDetect
+            GstAutoAudioSink
+            GstAutoAudioSrc
+            GstAutoVideoSink
+            GstAutoVideoSrc
           GstPipeline
             GstQTMoovRecover
           GstPushFileSrc
@@ -258,10 +263,11 @@ GObject
         GstRndBufferSize
         GstRtpJitterBuffer
         GstRtpPtDemux
+        GstRtpRtxReceive
+        GstRtpRtxSend
         GstRtpSession
         GstRtpSsrcDemux
-        GstRtpRtxSend
-        GstRtpRtxReceive
+        GstRtpStreamPay
         GstSMPTE
         GstShapeWipe
         GstTagDemux
@@ -275,10 +281,12 @@ GObject
           GstJpegDec
           GstPngDec
           GstVP8Dec
+          GstVP9Dec
         GstVideoEncoder
           GstJpegEnc
           GstPngEnc
           GstVP8Enc
+          GstVP9Enc
         GstVideoMixer2
         GstWavEnc
         GstWavParse
@@ -289,12 +297,14 @@ GObject
       GstPadTemplate
       GstPlugin
       GstPluginFeature
+        GstDeviceMonitorFactory
         GstElementFactory
         GstTypeFindFactory
       GstRegistry
       GstTask
       GstTaskPool
   GSocket
+  GTlsDatabase
   GdkPixbuf
   GstColorBalanceChannel
   GstTunerChannel
@@ -304,10 +314,10 @@ GObject
 GInterface
   GIcon
   GInitable
+  GLoadableIcon
   GTypePlugin
   GstChildProxy
   GstColorBalance
-  GstNavigation
   GstPreset
   GstStreamVolume
   GstTagSetter
@@ -316,4 +326,3 @@ GInterface
   GstTuner
   GstURIHandler
   GstVideoOrientation
-  GstVideoOverlay
diff --git a/docs/plugins/gst-plugins-good-plugins.interfaces b/docs/plugins/gst-plugins-good-plugins.interfaces
index e976d4cae..8dfb97d03 100644
--- a/docs/plugins/gst-plugins-good-plugins.interfaces
+++ b/docs/plugins/gst-plugins-good-plugins.interfaces
@@ -1,11 +1,13 @@
 GSocket GInitable
 GdkPixbuf GIcon
+GdkPixbuf GIcon GLoadableIcon
 Gst3GPPMux GstTagSetter GstTagXmpWriter
 GstApev2Mux GstTagSetter
 GstAspectRatioCrop GstChildProxy
 GstAudioEncoder GstPreset
 GstAutoAudioSink GstChildProxy
 GstAutoAudioSrc GstChildProxy
+GstAutoDetect GstChildProxy
 GstAutoVideoSink GstChildProxy
 GstAutoVideoSrc GstChildProxy
 GstAviMux GstTagSetter
@@ -78,10 +80,12 @@ GstUDPSrc GstURIHandler
 GstV4l2Radio GstURIHandler GstImplementsInterface GstTuner GstPropertyProbe
 GstV4l2Radio GstURIHandler GstTuner
 GstV4l2Sink GstImplementsInterface GstXOverlay GstNavigation GstColorBalance GstVideoOrientation GstPropertyProbe
+GstV4l2Sink GstTuner GstColorBalance GstVideoOrientation
 GstV4l2Sink GstTuner GstVideoOverlay GstNavigation GstColorBalance GstVideoOrientation
 GstV4l2Src GstURIHandler GstImplementsInterface GstTuner GstColorBalance GstVideoOrientation GstPropertyProbe
 GstV4l2Src GstURIHandler GstTuner GstColorBalance GstVideoOrientation
 GstVP8Enc GstPreset GstTagSetter
+GstVP9Enc GstPreset GstTagSetter
 GstVideoBalance GstColorBalance
 GstVideoBalance GstImplementsInterface GstColorBalance
 GstVideoEncoder GstPreset
diff --git a/docs/plugins/gst-plugins-good-plugins.prerequisites b/docs/plugins/gst-plugins-good-plugins.prerequisites
index ab859280e..962b020ec 100644
--- a/docs/plugins/gst-plugins-good-plugins.prerequisites
+++ b/docs/plugins/gst-plugins-good-plugins.prerequisites
@@ -1,5 +1,6 @@
 GIcon GObject
 GInitable GObject
+GLoadableIcon GIcon GObject
 GstChildProxy GObject
 GstChildProxy GstObject
 GstColorBalance GstImplementsInterface GstElement
diff --git a/docs/plugins/gst-plugins-good-plugins.signals b/docs/plugins/gst-plugins-good-plugins.signals
index 794622b8e..b14af19a3 100644
--- a/docs/plugins/gst-plugins-good-plugins.signals
+++ b/docs/plugins/gst-plugins-good-plugins.signals
@@ -479,6 +479,64 @@ guint  arg2
 </SIGNAL>
 
 <SIGNAL>
+<NAME>GstRtpBin::new-jitterbuffer</NAME>
+<RETURNS>void</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+GstElement *arg1
+guint  arg2
+guint  arg3
+</SIGNAL>
+
+<SIGNAL>
+<NAME>GstRtpBin::request-aux-receiver</NAME>
+<RETURNS>GstElement*</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint  arg1
+</SIGNAL>
+
+<SIGNAL>
+<NAME>GstRtpBin::request-aux-sender</NAME>
+<RETURNS>GstElement*</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint  arg1
+</SIGNAL>
+
+<SIGNAL>
+<NAME>GstRtpBin::request-rtcp-decoder</NAME>
+<RETURNS>GstElement*</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint  arg1
+</SIGNAL>
+
+<SIGNAL>
+<NAME>GstRtpBin::request-rtcp-encoder</NAME>
+<RETURNS>GstElement*</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint  arg1
+</SIGNAL>
+
+<SIGNAL>
+<NAME>GstRtpBin::request-rtp-decoder</NAME>
+<RETURNS>GstElement*</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint  arg1
+</SIGNAL>
+
+<SIGNAL>
+<NAME>GstRtpBin::request-rtp-encoder</NAME>
+<RETURNS>GstElement*</RETURNS>
+<FLAGS>l</FLAGS>
+GstRtpBin *gstrtpbin
+guint  arg1
+</SIGNAL>
+
+<SIGNAL>
 <NAME>GstRtpJitterBuffer::clear-pt-map</NAME>
 <RETURNS>void</RETURNS>
 <FLAGS>la</FLAGS>
@@ -715,3 +773,11 @@ guint  arg1
 GstCaps *arg2
 </SIGNAL>
 
+<SIGNAL>
+<NAME>GstRTSPSrc::new-manager</NAME>
+<RETURNS>void</RETURNS>
+<FLAGS>fc</FLAGS>
+GstRTSPSrc *gstrtspsrc
+GstElement *arg1
+</SIGNAL>
+
diff --git a/docs/plugins/inspect/plugin-1394.xml b/docs/plugins/inspect/plugin-1394.xml
index a2cf34118..4c60e18cd 100644
--- a/docs/plugins/inspect/plugin-1394.xml
+++ b/docs/plugins/inspect/plugin-1394.xml
@@ -3,7 +3,7 @@
   <description>Source for video data via IEEE1394 interface</description>
   <filename>../../ext/raw1394/.libs/libgst1394.so</filename>
   <basename>libgst1394.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-aasink.xml b/docs/plugins/inspect/plugin-aasink.xml
index e35a0962c..ab51066d7 100644
--- a/docs/plugins/inspect/plugin-aasink.xml
+++ b/docs/plugins/inspect/plugin-aasink.xml
@@ -3,7 +3,7 @@
   <description>ASCII Art video sink</description>
   <filename>../../ext/aalib/.libs/libgstaasink.so</filename>
   <basename>libgstaasink.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-alaw.xml b/docs/plugins/inspect/plugin-alaw.xml
index e80dc97e0..a32dbb81c 100644
--- a/docs/plugins/inspect/plugin-alaw.xml
+++ b/docs/plugins/inspect/plugin-alaw.xml
@@ -3,7 +3,7 @@
   <description>ALaw audio conversion routines</description>
   <filename>../../gst/law/.libs/libgstalaw.so</filename>
   <basename>libgstalaw.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-alpha.xml b/docs/plugins/inspect/plugin-alpha.xml
index 99e4811d3..7f6c5a0ab 100644
--- a/docs/plugins/inspect/plugin-alpha.xml
+++ b/docs/plugins/inspect/plugin-alpha.xml
@@ -3,7 +3,7 @@
   <description>adds an alpha channel to video - constant or via chroma-keying</description>
   <filename>../../gst/alpha/.libs/libgstalpha.so</filename>
   <basename>libgstalpha.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-alphacolor.xml b/docs/plugins/inspect/plugin-alphacolor.xml
index ebb423c05..02f284a68 100644
--- a/docs/plugins/inspect/plugin-alphacolor.xml
+++ b/docs/plugins/inspect/plugin-alphacolor.xml
@@ -3,7 +3,7 @@
   <description>RGBA from/to AYUV colorspace conversion preserving the alpha channel</description>
   <filename>../../gst/alpha/.libs/libgstalphacolor.so</filename>
   <basename>libgstalphacolor.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-apetag.xml b/docs/plugins/inspect/plugin-apetag.xml
index ef3d3d3e0..fb04308ec 100644
--- a/docs/plugins/inspect/plugin-apetag.xml
+++ b/docs/plugins/inspect/plugin-apetag.xml
@@ -3,7 +3,7 @@
   <description>APEv1/2 tag reader</description>
   <filename>../../gst/apetag/.libs/libgstapetag.so</filename>
   <basename>libgstapetag.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-audiofx.xml b/docs/plugins/inspect/plugin-audiofx.xml
index 7e526c828..2da24ab13 100644
--- a/docs/plugins/inspect/plugin-audiofx.xml
+++ b/docs/plugins/inspect/plugin-audiofx.xml
@@ -3,7 +3,7 @@
   <description>Audio effects plugin</description>
   <filename>../../gst/audiofx/.libs/libgstaudiofx.so</filename>
   <basename>libgstaudiofx.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-audioparsers.xml b/docs/plugins/inspect/plugin-audioparsers.xml
index f4a039df0..7a3bf28e2 100644
--- a/docs/plugins/inspect/plugin-audioparsers.xml
+++ b/docs/plugins/inspect/plugin-audioparsers.xml
@@ -3,7 +3,7 @@
   <description>Parsers for various audio formats</description>
   <filename>../../gst/audioparsers/.libs/libgstaudioparsers.so</filename>
   <basename>libgstaudioparsers.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-auparse.xml b/docs/plugins/inspect/plugin-auparse.xml
index c241d1d6e..677bfeabd 100644
--- a/docs/plugins/inspect/plugin-auparse.xml
+++ b/docs/plugins/inspect/plugin-auparse.xml
@@ -3,7 +3,7 @@
   <description>parses au streams</description>
   <filename>../../gst/auparse/.libs/libgstauparse.so</filename>
   <basename>libgstauparse.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-autodetect.xml b/docs/plugins/inspect/plugin-autodetect.xml
index 6cb3bd806..962b7074b 100644
--- a/docs/plugins/inspect/plugin-autodetect.xml
+++ b/docs/plugins/inspect/plugin-autodetect.xml
@@ -3,7 +3,7 @@
   <description>Plugin contains auto-detection plugins for video/audio in- and outputs</description>
   <filename>../../gst/autodetect/.libs/libgstautodetect.so</filename>
   <basename>libgstautodetect.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-avi.xml b/docs/plugins/inspect/plugin-avi.xml
index a95dbaa9b..f03c50a5c 100644
--- a/docs/plugins/inspect/plugin-avi.xml
+++ b/docs/plugins/inspect/plugin-avi.xml
@@ -3,7 +3,7 @@
   <description>AVI stream handling</description>
   <filename>../../gst/avi/.libs/libgstavi.so</filename>
   <basename>libgstavi.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -29,6 +29,12 @@
           <details>audio/ms-gsm, rate=(int)[ 1, 96000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)2, rate=(int)[ 16000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-raw, format=(string){ S8, U8, S16LE, U16LE, S24LE, U24LE, S32LE, U32LE }, layout=(string)interleaved, rate=(int)[ 1000, 192000 ], channels=(int)[ 1, 8 ]; audio/x-vorbis, rate=(int)[ 1000, 192000 ], channels=(int)[ 1, 2 ]; audio/x-ac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ]; audio/x-dts; audio/mpeg, mpegversion=(int)4, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ]; audio/x-alaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-mulaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-wms, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)microsoft, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)dvi, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-truespeech, rate=(int)8000, channels=(int)[ 1, 2 ]; audio/x-wma, wmaversion=(int)1, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ], block_align=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)2, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ], block_align=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)3, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ], block_align=(int)[ 1, 2147483647 ]; audio/x-vnd.sony.atrac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-raw, format=(string){ F32LE, F64LE }, layout=(string)interleaved, rate=(int)[ 1000, 192000 ], channels=(int)[ 1, 8 ]; audio/x-voxware, voxwaretype=(int)117, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-adpcm, layout=(string)dk4, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-adpcm, layout=(string)dk3, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-adpcm, layout=(string)dvi, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]; audio/AMR, rate=(int)8000, channels=(int)1; audio/AMR-WB, rate=(int)16000, channels=(int)1; audio/x-siren; application/x-ogg-avi; audio/x-avi-unknown</details>
         </caps>
         <caps>
+          <name>subpicture_%u</name>
+          <direction>source</direction>
+          <presence>sometimes</presence>
+          <details>subpicture/x-xsub</details>
+        </caps>
+        <caps>
           <name>subtitle_%u</name>
           <direction>source</direction>
           <presence>sometimes</presence>
diff --git a/docs/plugins/inspect/plugin-cacasink.xml b/docs/plugins/inspect/plugin-cacasink.xml
index 6ccd37391..7fa81fd0d 100644
--- a/docs/plugins/inspect/plugin-cacasink.xml
+++ b/docs/plugins/inspect/plugin-cacasink.xml
@@ -3,7 +3,7 @@
   <description>Colored ASCII Art video sink</description>
   <filename>../../ext/libcaca/.libs/libgstcacasink.so</filename>
   <basename>libgstcacasink.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-cairo.xml b/docs/plugins/inspect/plugin-cairo.xml
index 01d3c96ce..599998389 100644
--- a/docs/plugins/inspect/plugin-cairo.xml
+++ b/docs/plugins/inspect/plugin-cairo.xml
@@ -3,7 +3,7 @@
   <description>Cairo-based elements</description>
   <filename>../../ext/cairo/.libs/libgstcairo.so</filename>
   <basename>libgstcairo.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -20,13 +20,13 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, BGRA }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, BGRA, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ BGRx, BGRA }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ BGRx, BGRA, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-cutter.xml b/docs/plugins/inspect/plugin-cutter.xml
index 74ed6bc7b..cb9691a1a 100644
--- a/docs/plugins/inspect/plugin-cutter.xml
+++ b/docs/plugins/inspect/plugin-cutter.xml
@@ -3,7 +3,7 @@
   <description>Audio Cutter to split audio into non-silent bits</description>
   <filename>../../gst/cutter/.libs/libgstcutter.so</filename>
   <basename>libgstcutter.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -20,13 +20,13 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S8LE, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string)interleaved</details>
+          <details>audio/x-raw, format=(string){ S8, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string)interleaved</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S8LE, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string)interleaved</details>
+          <details>audio/x-raw, format=(string){ S8, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string)interleaved</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-debug.xml b/docs/plugins/inspect/plugin-debug.xml
index 28d50689a..3f845f46a 100644
--- a/docs/plugins/inspect/plugin-debug.xml
+++ b/docs/plugins/inspect/plugin-debug.xml
@@ -3,7 +3,7 @@
   <description>elements for testing and debugging</description>
   <filename>../../gst/debugutils/.libs/libgstdebug.so</filename>
   <basename>libgstdebug.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-deinterlace.xml b/docs/plugins/inspect/plugin-deinterlace.xml
index c39639a91..3e6d61f67 100644
--- a/docs/plugins/inspect/plugin-deinterlace.xml
+++ b/docs/plugins/inspect/plugin-deinterlace.xml
@@ -3,7 +3,7 @@
   <description>Deinterlacer</description>
   <filename>../../gst/deinterlace/.libs/libgstdeinterlace.so</filename>
   <basename>libgstdeinterlace.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -20,13 +20,13 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-dtmf.xml b/docs/plugins/inspect/plugin-dtmf.xml
index 67bcbea70..30d1c4fd1 100644
--- a/docs/plugins/inspect/plugin-dtmf.xml
+++ b/docs/plugins/inspect/plugin-dtmf.xml
@@ -3,7 +3,7 @@
   <description>DTMF plugins</description>
   <filename>../../gst/dtmf/.libs/libgstdtmf.so</filename>
   <basename>libgstdtmf.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-dv.xml b/docs/plugins/inspect/plugin-dv.xml
index 9c9f79e10..5a516fb98 100644
--- a/docs/plugins/inspect/plugin-dv.xml
+++ b/docs/plugins/inspect/plugin-dv.xml
@@ -3,7 +3,7 @@
   <description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
   <filename>../../ext/dv/.libs/libgstdv.so</filename>
   <basename>libgstdv.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-effectv.xml b/docs/plugins/inspect/plugin-effectv.xml
index 117e9b99e..a0c087d38 100644
--- a/docs/plugins/inspect/plugin-effectv.xml
+++ b/docs/plugins/inspect/plugin-effectv.xml
@@ -3,7 +3,7 @@
   <description>effect plugins from the effectv project</description>
   <filename>../../gst/effectv/.libs/libgsteffectv.so</filename>
   <basename>libgsteffectv.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-equalizer.xml b/docs/plugins/inspect/plugin-equalizer.xml
index b5ebba22f..013dafe4e 100644
--- a/docs/plugins/inspect/plugin-equalizer.xml
+++ b/docs/plugins/inspect/plugin-equalizer.xml
@@ -3,7 +3,7 @@
   <description>GStreamer audio equalizers</description>
   <filename>../../gst/equalizer/.libs/libgstequalizer.so</filename>
   <basename>libgstequalizer.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-flac.xml b/docs/plugins/inspect/plugin-flac.xml
index d643aa992..ab52b4f45 100644
--- a/docs/plugins/inspect/plugin-flac.xml
+++ b/docs/plugins/inspect/plugin-flac.xml
@@ -3,7 +3,7 @@
   <description>The FLAC Lossless compressor Codec</description>
   <filename>../../ext/flac/.libs/libgstflac.so</filename>
   <basename>libgstflac.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -26,7 +26,7 @@
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S8LE, S16LE, S24_32LE, S32LE }, layout=(string)interleaved, rate=(int)[ 1, 655350 ], channels=(int)[ 1, 8 ]</details>
+          <details>audio/x-raw, format=(string){ S8, S16LE, S24_32LE, S32LE }, layout=(string)interleaved, rate=(int)[ 1, 655350 ], channels=(int)[ 1, 8 ]</details>
         </caps>
       </pads>
     </element>
@@ -41,7 +41,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S8LE, S16LE, S24LE, S24_32LE }, layout=(string)interleaved, rate=(int)[ 1, 655350 ], channels=(int)[ 1, 8 ]</details>
+          <details>audio/x-raw, format=(string){ S24LE, S24_32LE, S16LE, S8 }, layout=(string)interleaved, rate=(int)[ 1, 655350 ], channels=(int)[ 1, 8 ]</details>
         </caps>
         <caps>
           <name>src</name>
diff --git a/docs/plugins/inspect/plugin-flv.xml b/docs/plugins/inspect/plugin-flv.xml
index 0db77dfef..f0dd215c9 100644
--- a/docs/plugins/inspect/plugin-flv.xml
+++ b/docs/plugins/inspect/plugin-flv.xml
@@ -3,7 +3,7 @@
   <description>FLV muxing and demuxing plugin</description>
   <filename>../../gst/flv/.libs/libgstflv.so</filename>
   <basename>libgstflv.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -47,7 +47,7 @@
           <name>audio</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }, parsed=(boolean)true; audio/mpeg, mpegversion=(int)2, framed=(boolean)true; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, framed=(boolean)true; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw, format=(string){ U8, S16LE }, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }</details>
+          <details>audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }, parsed=(boolean)true; audio/mpeg, mpegversion=(int){ 4, 2 }, stream-format=(string)raw; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw, format=(string){ U8, S16LE }, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }</details>
         </caps>
         <caps>
           <name>video</name>
diff --git a/docs/plugins/inspect/plugin-flxdec.xml b/docs/plugins/inspect/plugin-flxdec.xml
index b2bfe87f9..27602f259 100644
--- a/docs/plugins/inspect/plugin-flxdec.xml
+++ b/docs/plugins/inspect/plugin-flxdec.xml
@@ -3,7 +3,7 @@
   <description>FLC/FLI/FLX video decoder</description>
   <filename>../../gst/flx/.libs/libgstflxdec.so</filename>
   <basename>libgstflxdec.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-gdkpixbuf.xml b/docs/plugins/inspect/plugin-gdkpixbuf.xml
index 2f230b758..670429776 100644
--- a/docs/plugins/inspect/plugin-gdkpixbuf.xml
+++ b/docs/plugins/inspect/plugin-gdkpixbuf.xml
@@ -3,7 +3,7 @@
   <description>GdkPixbuf-based image decoder, overlay and sink</description>
   <filename>../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so</filename>
   <basename>libgstgdkpixbuf.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-goom.xml b/docs/plugins/inspect/plugin-goom.xml
index e4b544dbe..0728eff88 100644
--- a/docs/plugins/inspect/plugin-goom.xml
+++ b/docs/plugins/inspect/plugin-goom.xml
@@ -3,7 +3,7 @@
   <description>GOOM visualization filter</description>
   <filename>../../gst/goom/.libs/libgstgoom.so</filename>
   <basename>libgstgoom.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-goom2k1.xml b/docs/plugins/inspect/plugin-goom2k1.xml
index 7dfb7dff7..070ad509a 100644
--- a/docs/plugins/inspect/plugin-goom2k1.xml
+++ b/docs/plugins/inspect/plugin-goom2k1.xml
@@ -3,7 +3,7 @@
   <description>GOOM 2k1 visualization filter</description>
   <filename>../../gst/goom2k1/.libs/libgstgoom2k1.so</filename>
   <basename>libgstgoom2k1.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-icydemux.xml b/docs/plugins/inspect/plugin-icydemux.xml
index 53c071de5..f90866f95 100644
--- a/docs/plugins/inspect/plugin-icydemux.xml
+++ b/docs/plugins/inspect/plugin-icydemux.xml
@@ -3,7 +3,7 @@
   <description>Demux ICY tags from a stream</description>
   <filename>../../gst/icydemux/.libs/libgsticydemux.so</filename>
   <basename>libgsticydemux.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-id3demux.xml b/docs/plugins/inspect/plugin-id3demux.xml
index 387f6af15..1df22eb37 100644
--- a/docs/plugins/inspect/plugin-id3demux.xml
+++ b/docs/plugins/inspect/plugin-id3demux.xml
@@ -3,7 +3,7 @@
   <description>Demux ID3v1 and ID3v2 tags from a file</description>
   <filename>../../gst/id3demux/.libs/libgstid3demux.so</filename>
   <basename>libgstid3demux.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-imagefreeze.xml b/docs/plugins/inspect/plugin-imagefreeze.xml
index 7d5239ad1..895f22914 100644
--- a/docs/plugins/inspect/plugin-imagefreeze.xml
+++ b/docs/plugins/inspect/plugin-imagefreeze.xml
@@ -3,7 +3,7 @@
   <description>Still frame stream generator</description>
   <filename>../../gst/imagefreeze/.libs/libgstimagefreeze.so</filename>
   <basename>libgstimagefreeze.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-interleave.xml b/docs/plugins/inspect/plugin-interleave.xml
index 2efc3b238..eb688cbf4 100644
--- a/docs/plugins/inspect/plugin-interleave.xml
+++ b/docs/plugins/inspect/plugin-interleave.xml
@@ -3,7 +3,7 @@
   <description>Audio interleaver/deinterleaver</description>
   <filename>../../gst/interleave/.libs/libgstinterleave.so</filename>
   <basename>libgstinterleave.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-isomp4.xml b/docs/plugins/inspect/plugin-isomp4.xml
index 0eeec7866..937eecb85 100644
--- a/docs/plugins/inspect/plugin-isomp4.xml
+++ b/docs/plugins/inspect/plugin-isomp4.xml
@@ -3,7 +3,7 @@
   <description>ISO base media file format support (mp4, 3gpp, qt, mj2)</description>
   <filename>../../gst/isomp4/.libs/libgstisomp4.so</filename>
   <basename>libgstisomp4.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -23,10 +23,16 @@
           <details>audio/AMR, rate=(int)8000, channels=(int)[ 1, 2 ]; audio/AMR-WB, rate=(int)16000, channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]</details>
         </caps>
         <caps>
+          <name>subtitle_%u</name>
+          <direction>sink</direction>
+          <presence>request</presence>
+          <details>text/x-raw, format=(string)utf8</details>
+        </caps>
+        <caps>
           <name>video_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
         </caps>
         <caps>
           <name>src</name>
@@ -53,7 +59,7 @@
           <name>video_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
         </caps>
         <caps>
           <name>src</name>
@@ -80,7 +86,7 @@
           <name>video_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>image/x-j2c, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; image/x-jpc, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>image/x-j2c, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/x-jpc, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
         </caps>
         <caps>
           <name>src</name>
@@ -104,10 +110,16 @@
           <details>audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/x-alac, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]</details>
         </caps>
         <caps>
+          <name>subtitle_%u</name>
+          <direction>sink</direction>
+          <presence>request</presence>
+          <details>text/x-raw, format=(string)utf8</details>
+        </caps>
+        <caps>
           <name>video_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-mp4-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-mp4-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
         </caps>
         <caps>
           <name>src</name>
@@ -173,10 +185,16 @@
           <details>audio/x-raw, format=(string){ S8, U8 }, layout=(string)interleaved, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ S16LE, S16BE }, layout=(string)interleaved, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ S24LE, S24BE }, layout=(string)interleaved, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ S32LE, S32BE }, layout=(string)interleaved, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)dvi, block_align=(int)[ 64, 8096 ], channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-mulaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/AMR, rate=(int)8000, channels=(int)[ 1, 2 ]; audio/AMR-WB, rate=(int)16000, channels=(int)[ 1, 2 ]; audio/x-alac, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]</details>
         </caps>
         <caps>
+          <name>subtitle_%u</name>
+          <direction>sink</direction>
+          <presence>request</presence>
+          <details>text/x-raw, format=(string)utf8</details>
+        </caps>
+        <caps>
           <name>video_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>video/x-raw, format=(string){ RGB, UYVY }, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-svq, svqversion=(int)3, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-dv, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp8, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-dirac, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-qt-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>video/x-raw, format=(string){ RGB, UYVY }, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-svq, svqversion=(int)3, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp8, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dirac, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-qt-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
         </caps>
         <caps>
           <name>src</name>
diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml
index da09aa53d..1deb50a32 100644
--- a/docs/plugins/inspect/plugin-jack.xml
+++ b/docs/plugins/inspect/plugin-jack.xml
@@ -3,7 +3,7 @@
   <description>JACK audio elements</description>
   <filename>../../ext/jack/.libs/libgstjack.so</filename>
   <basename>libgstjack.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-jpeg.xml b/docs/plugins/inspect/plugin-jpeg.xml
index 42f48b8f1..c4412d2c8 100644
--- a/docs/plugins/inspect/plugin-jpeg.xml
+++ b/docs/plugins/inspect/plugin-jpeg.xml
@@ -3,7 +3,7 @@
   <description>JPeg plugin library</description>
   <filename>../../ext/jpeg/.libs/libgstjpeg.so</filename>
   <basename>libgstjpeg.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -20,7 +20,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>image/jpeg, width=(int)[ 1, 65535 ], height=(int)[ 1, 65535 ]</details>
+          <details>image/jpeg</details>
         </caps>
         <caps>
           <name>src</name>
diff --git a/docs/plugins/inspect/plugin-level.xml b/docs/plugins/inspect/plugin-level.xml
index 2c151da84..6d36d51c2 100644
--- a/docs/plugins/inspect/plugin-level.xml
+++ b/docs/plugins/inspect/plugin-level.xml
@@ -3,7 +3,7 @@
   <description>Audio level plugin</description>
   <filename>../../gst/level/.libs/libgstlevel.so</filename>
   <basename>libgstlevel.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-matroska.xml b/docs/plugins/inspect/plugin-matroska.xml
index d22a19cc1..93a70a464 100644
--- a/docs/plugins/inspect/plugin-matroska.xml
+++ b/docs/plugins/inspect/plugin-matroska.xml
@@ -3,7 +3,7 @@
   <description>Matroska and WebM stream handling</description>
   <filename>../../gst/matroska/.libs/libgstmatroska.so</filename>
   <basename>libgstmatroska.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -53,7 +53,7 @@
           <name>audio_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, stream-format=(string)raw, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-eac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-dts, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ U8, S16BE, S16LE, S24BE, S24LE, S32BE, S32LE, F32LE, F64LE }, layout=(string)interleaved, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]</details>
+          <details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, stream-format=(string)raw, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-eac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-dts, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-opus; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ U8, S16BE, S16LE, S24BE, S24LE, S32BE, S32LE, F32LE, F64LE }, layout=(string)interleaved, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-adpcm, layout=(string)dvi, block_align=(int)[ 64, 8192 ], channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-adpcm, layout=(string)g726, channels=(int)1, rate=(int)8000</details>
         </caps>
         <caps>
           <name>subtitle_%u</name>
@@ -65,7 +65,7 @@
           <name>video_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-huffyuv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-theora; video/x-dirac, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-pn-realvideo, rmversion=(int)[ 1, 4 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp8, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw, format=(string){ YUY2, I420, YV12, UYVY, AYUV }, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)[ 1, 3 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
+          <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, stream-format=(string)avc, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h265, stream-format=(string)hvc1, alignment=(string)au, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-huffyuv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-theora; video/x-dirac, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-pn-realvideo, rmversion=(int)[ 1, 4 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp8, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw, format=(string){ YUY2, I420, YV12, UYVY, AYUV, GRAY8, BGR, RGB }, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)[ 1, 3 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]</details>
         </caps>
         <caps>
           <name>src</name>
diff --git a/docs/plugins/inspect/plugin-mulaw.xml b/docs/plugins/inspect/plugin-mulaw.xml
index e94a41a15..5e0e28ead 100644
--- a/docs/plugins/inspect/plugin-mulaw.xml
+++ b/docs/plugins/inspect/plugin-mulaw.xml
@@ -3,7 +3,7 @@
   <description>MuLaw audio conversion routines</description>
   <filename>../../gst/law/.libs/libgstmulaw.so</filename>
   <basename>libgstmulaw.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-multifile.xml b/docs/plugins/inspect/plugin-multifile.xml
index d29285ef4..c4554e52c 100644
--- a/docs/plugins/inspect/plugin-multifile.xml
+++ b/docs/plugins/inspect/plugin-multifile.xml
@@ -3,7 +3,7 @@
   <description>Reads/Writes buffers from/to sequentially named files</description>
   <filename>../../gst/multifile/.libs/libgstmultifile.so</filename>
   <basename>libgstmultifile.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-multipart.xml b/docs/plugins/inspect/plugin-multipart.xml
index 6c88732ec..82dbf7fa1 100644
--- a/docs/plugins/inspect/plugin-multipart.xml
+++ b/docs/plugins/inspect/plugin-multipart.xml
@@ -3,7 +3,7 @@
   <description>multipart stream manipulation</description>
   <filename>../../gst/multipart/.libs/libgstmultipart.so</filename>
   <basename>libgstmultipart.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-navigationtest.xml b/docs/plugins/inspect/plugin-navigationtest.xml
index 94b92de37..0af9cd2d1 100644
--- a/docs/plugins/inspect/plugin-navigationtest.xml
+++ b/docs/plugins/inspect/plugin-navigationtest.xml
@@ -3,7 +3,7 @@
   <description>Template for a video filter</description>
   <filename>../../gst/debugutils/.libs/libgstnavigationtest.so</filename>
   <basename>libgstnavigationtest.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-oss4.xml b/docs/plugins/inspect/plugin-oss4.xml
index fdbcea471..ec1054138 100644
--- a/docs/plugins/inspect/plugin-oss4.xml
+++ b/docs/plugins/inspect/plugin-oss4.xml
@@ -3,7 +3,7 @@
   <description>Open Sound System (OSS) version 4 support for GStreamer</description>
   <filename>../../sys/oss4/.libs/libgstoss4audio.so</filename>
   <basename>libgstoss4audio.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-ossaudio.xml b/docs/plugins/inspect/plugin-ossaudio.xml
index 8ce5c5db8..499f3ed41 100644
--- a/docs/plugins/inspect/plugin-ossaudio.xml
+++ b/docs/plugins/inspect/plugin-ossaudio.xml
@@ -3,7 +3,7 @@
   <description>OSS (Open Sound System) support for GStreamer</description>
   <filename>../../sys/oss/.libs/libgstossaudio.so</filename>
   <basename>libgstossaudio.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-png.xml b/docs/plugins/inspect/plugin-png.xml
index a32b56c2a..ca0d4b39b 100644
--- a/docs/plugins/inspect/plugin-png.xml
+++ b/docs/plugins/inspect/plugin-png.xml
@@ -3,7 +3,7 @@
   <description>PNG plugin library</description>
   <filename>../../ext/libpng/.libs/libgstpng.so</filename>
   <basename>libgstpng.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-pulseaudio.xml b/docs/plugins/inspect/plugin-pulseaudio.xml
index 08d32083f..254ad244f 100644
--- a/docs/plugins/inspect/plugin-pulseaudio.xml
+++ b/docs/plugins/inspect/plugin-pulseaudio.xml
@@ -3,7 +3,7 @@
   <description>PulseAudio plugin library</description>
   <filename>../../ext/pulse/.libs/libgstpulse.so</filename>
   <basename>libgstpulse.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-replaygain.xml b/docs/plugins/inspect/plugin-replaygain.xml
index dc4c8bd38..a9255efde 100644
--- a/docs/plugins/inspect/plugin-replaygain.xml
+++ b/docs/plugins/inspect/plugin-replaygain.xml
@@ -3,7 +3,7 @@
   <description>ReplayGain volume normalization</description>
   <filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
   <basename>libgstreplaygain.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml
index e5c0e0bc6..e628c7b06 100644
--- a/docs/plugins/inspect/plugin-rtp.xml
+++ b/docs/plugins/inspect/plugin-rtp.xml
@@ -3,7 +3,7 @@
   <description>Real-time protocol plugins</description>
   <filename>../../gst/rtp/.libs/libgstrtp.so</filename>
   <basename>libgstrtp.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -167,7 +167,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1; application/x-rtp, media=(string)audio, clock-rate=(int)16000, encoding-name=(string)AMR-WB, encoding-params=(string)1, octet-align=(string)1</details>
+          <details>application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, octet-align=(string)1; application/x-rtp, media=(string)audio, clock-rate=(int)16000, encoding-name=(string)AMR-WB, octet-align=(string)1</details>
         </caps>
         <caps>
           <name>src</name>
@@ -713,7 +713,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(string){ 20, 30 }</details>
+          <details>application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)ILBC</details>
         </caps>
         <caps>
           <name>src</name>
@@ -1301,7 +1301,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>application/x-rtp, media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SIREN, dct-length=(int)320</details>
+          <details>application/x-rtp, media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SIREN</details>
         </caps>
         <caps>
           <name>src</name>
@@ -1343,7 +1343,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>application/x-rtp, media=(string)audio, clock-rate=(int)[ 6000, 48000 ], encoding-name=(string)SPEEX, encoding-params=(string)1</details>
+          <details>application/x-rtp, media=(string)audio, clock-rate=(int)[ 6000, 48000 ], encoding-name=(string)SPEEX</details>
         </caps>
         <caps>
           <name>src</name>
@@ -1375,6 +1375,48 @@
       </pads>
     </element>
     <element>
+      <name>rtpstreamdepay</name>
+      <longname>RTP Stream Depayloading</longname>
+      <class>Codec/Depayloader/Network</class>
+      <description>Depayloads RTP/RTCP packets for streaming protocols according to RFC4571</description>
+      <author>Sebastian Dröge &lt;sebastian@centricular.com&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>application/x-rtp-stream; application/x-rtcp-stream; application/x-srtp-stream; application/x-srtcp-stream</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>application/x-rtp; application/x-rtcp; application/x-srtp; application/x-srtcp</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
+      <name>rtpstreampay</name>
+      <longname>RTP Stream Payloading</longname>
+      <class>Codec/Payloader/Network</class>
+      <description>Payloads RTP/RTCP packets for streaming protocols according to RFC4571</description>
+      <author>Sebastian Dröge &lt;sebastian@centricular.com&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>application/x-rtp; application/x-rtcp; application/x-srtp; application/x-srtcp</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>application/x-rtp-stream; application/x-rtcp-stream; application/x-srtp-stream; application/x-srtcp-stream</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
       <name>rtpsv3vdepay</name>
       <longname>RTP SVQ3 depayloader</longname>
       <class>Codec/Depayloader/Network/RTP</class>
diff --git a/docs/plugins/inspect/plugin-rtpmanager.xml b/docs/plugins/inspect/plugin-rtpmanager.xml
index 79d41739b..4d4b69e03 100644
--- a/docs/plugins/inspect/plugin-rtpmanager.xml
+++ b/docs/plugins/inspect/plugin-rtpmanager.xml
@@ -3,7 +3,7 @@
   <description>RTP session management plugin library</description>
   <filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
   <basename>libgstrtpmanager.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -20,13 +20,13 @@
           <name>recv_rtcp_sink_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>application/x-rtcp</details>
+          <details>application/x-rtcp; application/x-srtcp</details>
         </caps>
         <caps>
           <name>recv_rtp_sink_%u</name>
           <direction>sink</direction>
           <presence>request</presence>
-          <details>application/x-rtp</details>
+          <details>application/x-rtp; application/x-srtp</details>
         </caps>
         <caps>
           <name>send_rtp_sink_%u</name>
@@ -44,13 +44,13 @@
           <name>send_rtp_src_%u</name>
           <direction>source</direction>
           <presence>sometimes</presence>
-          <details>application/x-rtp</details>
+          <details>application/x-rtp; application/x-srtp</details>
         </caps>
         <caps>
           <name>send_rtcp_src_%u</name>
           <direction>source</direction>
           <presence>request</presence>
-          <details>application/x-rtcp</details>
+          <details>application/x-rtcp; application/x-srtcp</details>
         </caps>
       </pads>
     </element>
@@ -172,6 +172,48 @@
       </pads>
     </element>
     <element>
+      <name>rtprtxreceive</name>
+      <longname>RTP Retransmission receiver</longname>
+      <class>Codec</class>
+      <description>Receive retransmitted RTP packets according to RFC4588</description>
+      <author>Julien Isorce &lt;julien.isorce@collabora.co.uk&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>application/x-rtp</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>application/x-rtp</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
+      <name>rtprtxsend</name>
+      <longname>RTP Retransmission Sender</longname>
+      <class>Codec</class>
+      <description>Retransmit RTP packets when needed, according to RFC4588</description>
+      <author>Julien Isorce &lt;julien.isorce@collabora.co.uk&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>application/x-rtp, clock-rate=(int)[ 1, 2147483647 ]</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>application/x-rtp</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
       <name>rtpsession</name>
       <longname>RTP Session</longname>
       <class>Filter/Network/RTP</class>
@@ -255,47 +297,5 @@
         </caps>
       </pads>
     </element>
-    <element>
-      <name>rtprtxsend</name>
-      <longname>RTP Retransmission Sender</longname>
-      <class>Codec</class>
-      <description>Retransmit RTP packets when needed, according to RFC4588</description>
-      <author>Julien Isorce &lt;julien.isorce@gmail.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>application/x-rtp</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>rtprtxreceive</name>
-      <longname>RTP Retransmission Receiver</longname>
-      <class>Codec</class>
-      <description>Receive retransmitted RTP packets according to RFC4588</description>
-      <author>Julien Isorce &lt;julien.isorce@gmail.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>application/x-rtp</details>
-        </caps>
-      </pads>
-    </element>
   </elements>
 </plugin>
\ No newline at end of file
diff --git a/docs/plugins/inspect/plugin-rtsp.xml b/docs/plugins/inspect/plugin-rtsp.xml
index 9ab755afe..431a76c6a 100644
--- a/docs/plugins/inspect/plugin-rtsp.xml
+++ b/docs/plugins/inspect/plugin-rtsp.xml
@@ -3,7 +3,7 @@
   <description>transfer data via RTSP</description>
   <filename>../../gst/rtsp/.libs/libgstrtsp.so</filename>
   <basename>libgstrtsp.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-shapewipe.xml b/docs/plugins/inspect/plugin-shapewipe.xml
index b2adf8f4a..ffaa86dbd 100644
--- a/docs/plugins/inspect/plugin-shapewipe.xml
+++ b/docs/plugins/inspect/plugin-shapewipe.xml
@@ -3,7 +3,7 @@
   <description>Shape Wipe transition filter</description>
   <filename>../../gst/shapewipe/.libs/libgstshapewipe.so</filename>
   <basename>libgstshapewipe.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-shout2send.xml b/docs/plugins/inspect/plugin-shout2send.xml
index 29a06cfee..ea7de39b9 100644
--- a/docs/plugins/inspect/plugin-shout2send.xml
+++ b/docs/plugins/inspect/plugin-shout2send.xml
@@ -3,7 +3,7 @@
   <description>Sends data to an icecast server using libshout2</description>
   <filename>../../ext/shout2/.libs/libgstshout2.so</filename>
   <basename>libgstshout2.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>libshout2</package>
diff --git a/docs/plugins/inspect/plugin-smpte.xml b/docs/plugins/inspect/plugin-smpte.xml
index f919a265d..880784c5d 100644
--- a/docs/plugins/inspect/plugin-smpte.xml
+++ b/docs/plugins/inspect/plugin-smpte.xml
@@ -3,7 +3,7 @@
   <description>Apply the standard SMPTE transitions on video images</description>
   <filename>../../gst/smpte/.libs/libgstsmpte.so</filename>
   <basename>libgstsmpte.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-soup.xml b/docs/plugins/inspect/plugin-soup.xml
index d42f0169e..57025b426 100644
--- a/docs/plugins/inspect/plugin-soup.xml
+++ b/docs/plugins/inspect/plugin-soup.xml
@@ -3,7 +3,7 @@
   <description>libsoup HTTP client src/sink</description>
   <filename>../../ext/soup/.libs/libgstsouphttpsrc.so</filename>
   <basename>libgstsouphttpsrc.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-spectrum.xml b/docs/plugins/inspect/plugin-spectrum.xml
index c50bf172c..731eeae79 100644
--- a/docs/plugins/inspect/plugin-spectrum.xml
+++ b/docs/plugins/inspect/plugin-spectrum.xml
@@ -3,7 +3,7 @@
   <description>Run an FFT on the audio signal, output spectrum data</description>
   <filename>../../gst/spectrum/.libs/libgstspectrum.so</filename>
   <basename>libgstspectrum.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-speex.xml b/docs/plugins/inspect/plugin-speex.xml
index ce8c52603..f782108d6 100644
--- a/docs/plugins/inspect/plugin-speex.xml
+++ b/docs/plugins/inspect/plugin-speex.xml
@@ -3,7 +3,7 @@
   <description>Speex plugin library</description>
   <filename>../../ext/speex/.libs/libgstspeex.so</filename>
   <basename>libgstspeex.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-taglib.xml b/docs/plugins/inspect/plugin-taglib.xml
index a70983cd8..f8bf15c81 100644
--- a/docs/plugins/inspect/plugin-taglib.xml
+++ b/docs/plugins/inspect/plugin-taglib.xml
@@ -3,7 +3,7 @@
   <description>Tag writing plug-in based on taglib</description>
   <filename>../../ext/taglib/.libs/libgsttaglib.so</filename>
   <basename>libgsttaglib.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-udp.xml b/docs/plugins/inspect/plugin-udp.xml
index cff3b4885..114731805 100644
--- a/docs/plugins/inspect/plugin-udp.xml
+++ b/docs/plugins/inspect/plugin-udp.xml
@@ -3,7 +3,7 @@
   <description>transfer data via UDP</description>
   <filename>../../gst/udp/.libs/libgstudp.so</filename>
   <basename>libgstudp.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-video4linux2.xml b/docs/plugins/inspect/plugin-video4linux2.xml
index 349a008d4..e74cc81fc 100644
--- a/docs/plugins/inspect/plugin-video4linux2.xml
+++ b/docs/plugins/inspect/plugin-video4linux2.xml
@@ -3,7 +3,7 @@
   <description>elements for Video 4 Linux</description>
   <filename>../../sys/v4l2/.libs/libgstvideo4linux2.so</filename>
   <basename>libgstvideo4linux2.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -29,7 +29,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string)RGB15, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)RGB16, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)BGR, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)RGB, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)BGRx, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)RGBx, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)GRAY8, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YVU9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YUY2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)UYVY, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)Y42B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)Y41B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)NV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)NV21, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YUV9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)I420, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/mpegts; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-h263, variant=(string)itu, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-h264, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YVYU, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]</details>
+          <details>image/jpeg; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; video/mpeg, mpegversion=(int)2; video/mpegts, systemstream=(boolean)true; video/x-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true; video/x-h263, variant=(string)itu; video/x-h264, stream-format=(string)byte-stream, alignment=(string)au; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string){ RGB15, RGB16, BGR, RGB, BGRx, xRGB, GRAY8, YVU9, YV12, YUY2, UYVY, Y42B, Y41B, NV12_64Z32, YUV9, I420, YVYU, NV21, NV12 }, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-vp8</details>
         </caps>
       </pads>
     </element>
@@ -44,7 +44,7 @@
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>video/x-raw, format=(string)RGB15, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)RGB16, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)BGR, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)RGB, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)BGRx, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)RGBx, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)GRAY8, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YVU9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YUY2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)UYVY, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)Y42B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)Y41B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)NV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)NV21, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YUV9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)I420, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/mpegts; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-h263, variant=(string)itu, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-h264, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string)YVYU, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]</details>
+          <details>image/jpeg; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; video/mpeg, mpegversion=(int)2; video/mpegts, systemstream=(boolean)true; video/x-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true; video/x-h263, variant=(string)itu; video/x-h264, stream-format=(string)byte-stream, alignment=(string)au; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw, format=(string){ RGB15, RGB16, BGR, RGB, BGRx, xRGB, GRAY8, YVU9, YV12, YUY2, UYVY, Y42B, Y41B, NV12_64Z32, YUV9, I420, YVYU, NV21, NV12 }, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-vp8</details>
         </caps>
       </pads>
     </element>
diff --git a/docs/plugins/inspect/plugin-videobox.xml b/docs/plugins/inspect/plugin-videobox.xml
index 8603c773a..490f960b6 100644
--- a/docs/plugins/inspect/plugin-videobox.xml
+++ b/docs/plugins/inspect/plugin-videobox.xml
@@ -3,7 +3,7 @@
   <description>resizes a video by adding borders or cropping</description>
   <filename>../../gst/videobox/.libs/libgstvideobox.so</filename>
   <basename>libgstvideobox.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-videocrop.xml b/docs/plugins/inspect/plugin-videocrop.xml
index c2ea1ebba..f213c189b 100644
--- a/docs/plugins/inspect/plugin-videocrop.xml
+++ b/docs/plugins/inspect/plugin-videocrop.xml
@@ -3,7 +3,7 @@
   <description>Crops video into a user-defined region</description>
   <filename>../../gst/videocrop/.libs/libgstvideocrop.so</filename>
   <basename>libgstvideocrop.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-videofilter.xml b/docs/plugins/inspect/plugin-videofilter.xml
index 99668a26c..a58165e89 100644
--- a/docs/plugins/inspect/plugin-videofilter.xml
+++ b/docs/plugins/inspect/plugin-videofilter.xml
@@ -3,7 +3,7 @@
   <description>Video filters plugin</description>
   <filename>../../gst/videofilter/.libs/libgstvideofilter.so</filename>
   <basename>libgstvideofilter.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-videomixer.xml b/docs/plugins/inspect/plugin-videomixer.xml
index 250d64752..0526574dc 100644
--- a/docs/plugins/inspect/plugin-videomixer.xml
+++ b/docs/plugins/inspect/plugin-videomixer.xml
@@ -3,7 +3,7 @@
   <description>Video mixer</description>
   <filename>../../gst/videomixer/.libs/libgstvideomixer.so</filename>
   <basename>libgstvideomixer.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-vpx.xml b/docs/plugins/inspect/plugin-vpx.xml
index 12b062866..7338cb6ac 100644
--- a/docs/plugins/inspect/plugin-vpx.xml
+++ b/docs/plugins/inspect/plugin-vpx.xml
@@ -3,7 +3,7 @@
   <description>VP8 plugin</description>
   <filename>../../ext/vpx/.libs/libgstvpx.so</filename>
   <basename>libgstvpx.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -51,5 +51,47 @@
         </caps>
       </pads>
     </element>
+    <element>
+      <name>vp9dec</name>
+      <longname>On2 VP9 Decoder</longname>
+      <class>Codec/Decoder/Video</class>
+      <description>Decode VP9 video streams</description>
+      <author>David Schleef &lt;ds@entropywave.com&gt;, Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>video/x-vp9</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>video/x-raw, format=(string){ I420, YV12, Y42B, Y444 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
+      <name>vp9enc</name>
+      <longname>On2 VP9 Encoder</longname>
+      <class>Codec/Encoder/Video</class>
+      <description>Encode VP9 video streams</description>
+      <author>David Schleef &lt;ds@entropywave.com&gt;, Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>video/x-raw, format=(string){ I420, YV12 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>video/x-vp9, profile=(string){ 0, 1, 2, 3 }</details>
+        </caps>
+      </pads>
+    </element>
   </elements>
 </plugin>
\ No newline at end of file
diff --git a/docs/plugins/inspect/plugin-wavenc.xml b/docs/plugins/inspect/plugin-wavenc.xml
index 68cc83ab4..59d0e4b57 100644
--- a/docs/plugins/inspect/plugin-wavenc.xml
+++ b/docs/plugins/inspect/plugin-wavenc.xml
@@ -3,7 +3,7 @@
   <description>Encode raw audio into WAV</description>
   <filename>../../gst/wavenc/.libs/libgstwavenc.so</filename>
   <basename>libgstwavenc.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
@@ -20,7 +20,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)1, format=(string){ S32LE, S24LE, S16LE, U8, F32LE, F64LE }, layout=(string)interleaved; audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)2, channel-mask=(bitmask)0x0000000000000003, format=(string){ S32LE, S24LE, S16LE, U8, F32LE, F64LE }, layout=(string)interleaved; audio/x-alaw, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ], width=(int)8, depth=(int)8, signed=(boolean)false; audio/x-mulaw, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ], width=(int)8, depth=(int)8, signed=(boolean)false</details>
+          <details>audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)1, format=(string){ S32LE, S24LE, S16LE, U8, F32LE, F64LE }, layout=(string)interleaved; audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)2, channel-mask=(bitmask)0x0000000000000003, format=(string){ S32LE, S24LE, S16LE, U8, F32LE, F64LE }, layout=(string)interleaved; audio/x-alaw, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ]; audio/x-mulaw, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ]</details>
         </caps>
         <caps>
           <name>src</name>
diff --git a/docs/plugins/inspect/plugin-wavpack.xml b/docs/plugins/inspect/plugin-wavpack.xml
index ff91e1a66..8a22c04f7 100644
--- a/docs/plugins/inspect/plugin-wavpack.xml
+++ b/docs/plugins/inspect/plugin-wavpack.xml
@@ -3,7 +3,7 @@
   <description>Wavpack lossless/lossy audio format handling</description>
   <filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
   <basename>libgstwavpack.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-wavparse.xml b/docs/plugins/inspect/plugin-wavparse.xml
index db62d993e..b6784d564 100644
--- a/docs/plugins/inspect/plugin-wavparse.xml
+++ b/docs/plugins/inspect/plugin-wavparse.xml
@@ -3,7 +3,7 @@
   <description>Parse a .wav file into raw audio</description>
   <filename>../../gst/wavparse/.libs/libgstwavparse.so</filename>
   <basename>libgstwavparse.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-ximagesrc.xml b/docs/plugins/inspect/plugin-ximagesrc.xml
index 589a07274..4b240e184 100644
--- a/docs/plugins/inspect/plugin-ximagesrc.xml
+++ b/docs/plugins/inspect/plugin-ximagesrc.xml
@@ -3,7 +3,7 @@
   <description>X11 video input plugin using standard Xlib calls</description>
   <filename>../../sys/ximage/.libs/libgstximagesrc.so</filename>
   <basename>libgstximagesrc.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/docs/plugins/inspect/plugin-y4menc.xml b/docs/plugins/inspect/plugin-y4menc.xml
index 089c1053f..e7359a241 100644
--- a/docs/plugins/inspect/plugin-y4menc.xml
+++ b/docs/plugins/inspect/plugin-y4menc.xml
@@ -3,7 +3,7 @@
   <description>Encodes a YUV frame into the yuv4mpeg format (mjpegtools)</description>
   <filename>../../gst/y4m/.libs/libgsty4menc.so</filename>
   <basename>libgsty4menc.so</basename>
-  <version>1.2.0</version>
+  <version>1.3.1</version>
   <license>LGPL</license>
   <source>gst-plugins-good</source>
   <package>GStreamer Good Plug-ins source release</package>
diff --git a/gst-plugins-good.doap b/gst-plugins-good.doap
index b28dbe507..460ffae99 100644
--- a/gst-plugins-good.doap
+++ b/gst-plugins-good.doap
@@ -34,6 +34,16 @@ the plug-in code, LGPL or LGPL-compatible for the supporting library).
 
  <release>
   <Version>
+   <revision>1.3.1</revision>
+   <branch>1.3</branch>
+   <name></name>
+   <created>2014-05-03</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.3.1.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.2.0</revision>
    <branch>1.2</branch>
    <name></name>
diff --git a/gst/audiofx/audiopanoramaorc-dist.c b/gst/audiofx/audiopanoramaorc-dist.c
index 438cc4047..3d5999a61 100644
--- a/gst/audiofx/audiopanoramaorc-dist.c
+++ b/gst/audiofx/audiopanoramaorc-dist.c
@@ -156,8 +156,8 @@ void audiopanoramam_orc_process_f32_ch2_sim_left (gfloat * ORC_RESTRICT d1,
 #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX)
 #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX)
 #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX)
-#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8))
-#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24))
+#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8))
+#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24))
 #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56))
 #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset)))
 #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff))
diff --git a/gst/deinterlace/tvtime-dist.c b/gst/deinterlace/tvtime-dist.c
index fab309064..b121c010a 100644
--- a/gst/deinterlace/tvtime-dist.c
+++ b/gst/deinterlace/tvtime-dist.c
@@ -133,8 +133,8 @@ void deinterlace_line_greedy (orc_uint8 * ORC_RESTRICT d1,
 #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX)
 #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX)
 #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX)
-#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8))
-#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24))
+#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8))
+#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24))
 #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56))
 #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset)))
 #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff))
diff --git a/gst/videobox/gstvideoboxorc-dist.c b/gst/videobox/gstvideoboxorc-dist.c
index 15bbf1def..8fef7a049 100644
--- a/gst/videobox/gstvideoboxorc-dist.c
+++ b/gst/videobox/gstvideoboxorc-dist.c
@@ -121,8 +121,8 @@ void video_box_orc_splat_u32 (guint32 * ORC_RESTRICT d1, int p1, int n);
 #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX)
 #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX)
 #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX)
-#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8))
-#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24))
+#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8))
+#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24))
 #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56))
 #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset)))
 #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff))
diff --git a/gst/videomixer/videomixerorc-dist.c b/gst/videomixer/videomixerorc-dist.c
index 9106640f5..05c186377 100644
--- a/gst/videomixer/videomixerorc-dist.c
+++ b/gst/videomixer/videomixerorc-dist.c
@@ -318,8 +318,8 @@ void videomixer_video_convert_orc_putline_A420 (guint8 * ORC_RESTRICT d1,
 #define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX)
 #define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX)
 #define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX)
-#define ORC_SWAP_W(x) ((((x)&0xff)<<8) | (((x)&0xff00)>>8))
-#define ORC_SWAP_L(x) ((((x)&0xff)<<24) | (((x)&0xff00)<<8) | (((x)&0xff0000)>>8) | (((x)&0xff000000)>>24))
+#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8))
+#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24))
 #define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56))
 #define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset)))
 #define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff))
diff --git a/win32/common/config.h b/win32/common/config.h
index cf4c3761f..1728c6a3e 100644
--- a/win32/common/config.h
+++ b/win32/common/config.h
@@ -58,17 +58,11 @@
 #define GST_PACKAGE_ORIGIN "Unknown package origin"
 
 /* GStreamer package release date/time for plugins as YYYY-MM-DD */
-#define GST_PACKAGE_RELEASE_DATETIME "2013-09-24"
+#define GST_PACKAGE_RELEASE_DATETIME "2014-05-03"
 
 /* Define if static plugins should be built */
 #undef GST_PLUGIN_BUILD_STATIC
 
-/* struct v4l2_buffer missing */
-#undef GST_V4L2_MISSING_BUFDECL
-
-/* I know the API is subject to change. */
-#undef G_UDEV_API_IS_SUBJECT_TO_CHANGE
-
 /* Define to enable aalib ASCII Art library (used by aasink). */
 #undef HAVE_AALIB
 
@@ -142,10 +136,6 @@
    */
 #undef HAVE_DCGETTEXT
 
-/* Define to 1 if you have the declaration of `V4L2_MEMORY_DMABUF', and to 0
-   if you don't. */
-#undef HAVE_DECL_V4L2_MEMORY_DMABUF
-
 /* Define to enable DirectSound plug-in (used by directsoundsink). */
 #undef HAVE_DIRECTSOUND
 
@@ -191,7 +181,7 @@
 /* Define if the GNU gettext() function is already present or preinstalled. */
 #undef HAVE_GETTEXT
 
-/* Define to enable Video 4 Linux 2 (used by v4l2src). */
+/* Define to enable Video 4 Linux 2 (used by video4linux2). */
 #undef HAVE_GST_V4L2
 
 /* Whether gudev is available for device detection */
@@ -359,9 +349,6 @@
 /* Define to enable X Shared Memory extension. */
 #undef HAVE_XSHM
 
-/* Define to enable X11 XVideo extensions. */
-#undef HAVE_XVIDEO
-
 /* Define to enable zlib support for qtdemux/matroska. */
 #undef HAVE_ZLIB
 
@@ -385,7 +372,7 @@
 #define PACKAGE_NAME "GStreamer Good Plug-ins"
 
 /* Define to the full name and version of this package. */
-#define PACKAGE_STRING "GStreamer Good Plug-ins 1.2.0"
+#define PACKAGE_STRING "GStreamer Good Plug-ins 1.3.1"
 
 /* Define to the one symbol short name of this package. */
 #define PACKAGE_TARNAME "gst-plugins-good"
@@ -394,7 +381,7 @@
 #undef PACKAGE_URL
 
 /* Define to the version of this package. */
-#define PACKAGE_VERSION "1.2.0"
+#define PACKAGE_VERSION "1.3.1"
 
 /* directory where plugins are located */
 #ifdef _DEBUG
@@ -428,7 +415,7 @@
 #undef TARGET_CPU
 
 /* Version number of package */
-#define VERSION "1.2.0"
+#define VERSION "1.3.1"
 
 /* old wavpack API */
 #undef WAVPACK_OLD_API