+=== 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>
projects / platform / upstream / gstreamer.git / commitdiff