Commit 17fd3fa8 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.2.1

parent dacce6e3
=== release 1.2.1 ===
2013-11-09 Sebastian Dröge <>
releasing 1.2.1
2013-11-09 12:01:55 +0100 Sebastian Dröge <>
* po/de.po:
* po/id.po:
* po/sr.po:
po: Update translations
2013-11-08 17:59:24 +0100 Philippe Normand <>
* 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.
2013-10-07 14:27:21 -0300 Thiago Santos <>
* 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.
2013-10-25 11:30:36 -0300 Thiago Santos <>
* 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.
2013-10-25 18:22:00 -0300 Thiago Santos <>
* 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.
2013-10-25 11:42:37 -0300 Thiago Santos <>
* 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.
2013-11-07 09:49:55 +0100 Sebastian Dröge <>
* 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 <>
* sys/osxaudio/gstosxcoreaudiohal.c:
osxaudiosink: fix segfault when we can't get the channels layout
2013-11-05 17:26:49 +0100 Sebastian Dröge <>
* 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.
2013-11-02 22:50:47 +0100 Rico Tzschichholz <>
* gst/rtsp/
* gst/rtsp/gstrtspsrc.h:
rtsp: Add missing gio-2.0 deps and includes
2013-11-01 18:10:51 +0000 Olivier Crête <>
Revert "configure: Require gst-plugins-base >= 1.2.1 for the TLS validation check flags in GstRTSPConnection"
Version 1.2.1 doesn't exist yet, re-apply when it does
This reverts commit c98380985db3483ea78a8e738d544d1201d8ed1e.
2013-11-01 18:31:36 +0100 Sebastian Dröge <>
* gst/audiofx/audioiirfilter.c:
audioiirfilter: Fix initialization coefficient handling
Broke unit test.
2013-11-01 16:59:11 +0100 Sebastian Dröge <>
configure: Require gst-plugins-base >= 1.2.1 for the TLS validation check flags in GstRTSPConnection
2013-10-31 14:05:43 -0700 Aleix Conchillo Flaque <>
* 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.
2013-10-31 22:43:49 +0100 Sebastian Dröge <>
* 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.
2013-10-31 19:15:12 +0100 Sebastian Dröge <>
* ext/wavpack/gstwavpackenc.c:
wavpackenc: Fix writing of MD5 sums and other metadata blocks
These don't have the FINAL_BLOCK flag set.
2013-10-14 16:23:25 +0200 Ognyan Tonchev <>
* gst/udp/gstmultiudpsink.c:
multiudpsink: Fix memory leak
Unmap all GstMemory of the current buffer when flushing.
2013-10-12 20:37:41 +0100 Tim-Philipp Müller <>
* 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
2013-10-10 13:52:35 +0200 Sebastian Dröge <>
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdec.h:
dvdec: Don't send segment event before caps
2013-10-09 17:46:33 +0200 Sebastian Dröge <>
* ext/dv/gstdvdemux.c:
dvdemux: Send stream-start, caps and segment events in the right order
2013-10-08 11:28:04 +0200 Sebastian Dröge <>
* gst/wavenc/gstwavenc.c:
wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields
2013-10-07 12:54:11 +0200 Sebastian Dröge <>
* gst/deinterlace/tvtime/greedyh.c:
deinterlace: Fix handling of planar video formats in greedyh method
2013-10-04 13:34:09 +0200 Peter Korsgaard <>
* 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.
2013-10-04 14:42:59 -0700 Reynaldo H. Verdejo Pinochet <>
* 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
2013-10-03 22:38:43 +0200 Mathieu Duponchelle <>
* gst/videomixer/videoconvert.c:
videomixer: Update videoconvert copy
2013-10-03 21:36:34 +0200 Mathieu Duponchelle <>
* gst/videomixer/videomixer2.c:
videomixer: Check if the pad needs reconfiguration in collected
2013-10-03 11:59:25 +0200 Sebastian Dröge <>
* gst/isomp4/qtdemux.c:
qtdemux: Add support for the mp2v fourcc for MPEG-2 video
2013-10-04 12:11:56 +0200 Wim Taymans <>
* 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
2013-10-03 14:39:35 +0100 Matthieu Bouron <>
* 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.
2013-10-02 15:56:53 +0200 Ognyan Tonchev <>
* gst/matroska/matroska-demux.c:
matroskademux: Fix memory leak
2013-09-30 12:24:32 +0200 Ognyan Tonchev <>
* gst/rtp/gstrtpgstpay.c:
rtpgstpay: Fix memory leak
We were leaking the GList nodes of the pending buffers.
2013-09-30 12:31:00 +0200 Wim Taymans <>
* 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 <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: small debug improvement
2013-09-30 11:53:08 +0200 Wim Taymans <>
* 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 <>
* 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 <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: improve debug
2013-09-27 15:05:04 +0200 Wim Taymans <>
* 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.
2013-09-26 16:20:04 +0200 Wim Taymans <>
* 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 17:42:02 +0200 Wim Taymans <>
* 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
2013-09-25 17:36:15 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpbin.c:
rtpbin: avoid some pad link checks
Link pads without checks, we know it will work.
2013-09-24 04:02:09 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: calculate some stats
2013-09-23 17:05:44 +0200 Wim Taymans <>
* 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.
2013-10-03 18:33:01 +0100 Tim-Philipp Müller <>
* sys/v4l2/gstv4l2object.c:
v4l2src: print probed caps as caps again in debug log
This got lost during refactoring.
2013-09-26 20:41:26 +0200 Hans Månsson <>
* gst/isomp4/gstqtmuxmap.c:
mp4mux: Do not require framerate in peer video caps
Remove the framerate restriction on the caps.
2013-09-16 11:20:51 -0300 Thiago Santos <>
* 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.
2013-09-21 00:55:26 +0200 Matej Knopp <>
* gst/matroska/matroska-demux.c:
matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily
2013-09-27 12:53:06 +0200 Matej Knopp <>
* gst/matroska/matroska-demux.c:
matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8
2013-09-25 12:55:21 +0200 Sebastian Dröge <>
* 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 17:24:26 +0100 Tim-Philipp Müller <>
* common:
Automatic update of common submodule
From 6b03ba7 to 7412249
=== release 1.2.0 ===
2013-09-24 Sebastian Dröge <>
2013-09-24 14:21:08 +0200 Sebastian Dröge <>
* ChangeLog:
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 <>
* 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 <>
This is GStreamer Good Plugins 1.2.0
Changes since 1.0:
New API:
• GstContext negotiation / sharing / announcing for sharing a
generic context between elements, e.g. a display handle
• GL texture upload conversion meta for allowing different
buffer types to be converted to an OpenGL texture
• GstCapsFeatures as extension to GstCaps for allowing the
negotiation of specific memory or meta requirements between
• GstMemory flags for contiguous and non-mappable memory
• The stream-start event has optional flags now, e.g. for signalling
sparse streams
• The stream-start even has an optional group-id field now to signal
all streams that should be played together
• Allocators library in gst-plugins-base, currently only with generic
dmabuf memory support
• insertbin library for easier handling of dynamically linked
pipelines (in -bad for now)
• EGL helper library (in -bad for now)
• MPEG-TS data structure library (in -bad for now)
• New GstVideoRegionOfInterestMeta to describe a region of interest on
video frames.
• GstVideoDecoder/Encoder has new ::flush() vfunc to replace the
ill-defined ::reset() vfunc.
• The URI query allows to query the redirected URI now.
Major changes:
• New tool: gst-play-1.0 in gst-plugins-base for basic playback
testing on the command line.
• New plugins:
∘ mssdemux for Microsoft Smooth Streaming
∘ dashdemux for DASH adaptive streaming protocol
∘ bluez for interaction with Bluetooth devices
∘ openjpeg for JPEG2000 decoding and encoding
∘ daala for experimental Daala decoding and encoding
∘ vpx plugin has experimental VP9 decoding and encoding support
∘ webp plugin for WebP decoding (encoding to be added later)
∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse,
mfc, ivtv, accuraterip and audiofxbad
• Moved plugins:
∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in
gst-plugins-good now
• Video:
∘ Fix handling of interlaced video in converters such as videoscale
and videoconvert (e.g. scale both fields independently)
∘ videoconvert will try harder to minimise quality losses when
conversion is necessary
∘ The experimental GstSurfaceConverter, GstSurfaceMeta and
GstVideoContext APIs from the (confusingly-named)
libgstbasevideo-1.0 library in gst-plugins-bad have now been
removed and been replaced by new APIs in GStreamer Core and
gst-plugins-base (see above). Since that was all that was left in
this library, the entire experimental libgstbasevideo-1.0 library
has been removed from gst-plugins-bad
∘ Chroma subsampling and chroma siting conversion is better handled
in videoconvert and the support for interlaced video was improved.
∘ New pinwheel and spoke patterns in videotestsrc
∘ videomixer can now accept different video formats on its sinkpads
and converts to a common format during mixing
• Audio:
∘ audioconvert will try harder to minimise quality losses when
conversion is necessary
∘ adder now allows muting/unmuting of its input streams, and also
per-input stream volume
∘ pulseaudio elements can switch between devices during playback now
∘ aacparse can convert between ADTS←→RAW
• Platform specific changes:
∘ Caps, events, etc. are now printed in the GStreamer debug logs
with their content instead of just the pointer address even on
non-glibc platforms (e.g. Windows, OSX, Android).
∘ Network elements (UDP/TCP) now work better with platforms,
where IPv6 sockets can't handle IPv4 (e.g. Windows)
∘ Linux/BSD: v4l2 had many improvements and cleanups
• Other changes:
∘ gst-libav now uses libav 9
∘ Static linking of plugins is supported now (also in 1.0.7)
∘ rtspsrc: add support for NetClientClock: when the server suggests a
GstNetTimeProvider in the SDP, set up a GstNetClientClock that
slaves to the remote clock and suggest this clock in provide_clock.
Simplifies synchronized playback of a resource from an RTSP server.
gst-rtsp-server now supports adding this to the SDP and can provide
a network clock
∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements
∘ SRTP and DTLS support
∘ Changes to many elements and core to use the correct sticky event
order and also not lose any important sticky events during flushing
∘ >1000 fixed bug reports, and many other bug fixes and other
improvements everywhere that had no bug report
Things to look out for:
• Single header includes for all libraries, e.g. #include
<gst/video/video.h> - this was needed for some bindings.
• Stricter (correct) caps subset checking in some cases where this was
not correct before. Caps will now always fail to be a compatible
subset of another set of caps if the subset caps are missing some
fields that the superset caps have. This might lead to not-negotiated
errors if caps are incomplete now. However, it also prevents possible
data corruption caused by piping data formatted in an
incompatible/unexpected way into some elements. Check your h264 caps
for stream-format and alignment fields and AAC caps for the
stream-format field. This change will also be included in the next
stable 1.0.8 release.
• Stricter checking for missing events and correct sticky event order
(stream-start, caps, segment) in some places; this is not enabled in
stable releases by default, but you may get warnings when using git
builds, development releases or when compiling with
• x264enc now outputs data in byte-stream by default if downstream has
ANY caps (e.g. appsink without caps set, filesink, udpsink,
tcpserversink etc.)
• The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a
different format now. This new format uses the data structures from
the new MPEGTS library
• The GstContext API has changed between 1.1.4 and 1.1.90
This is GStreamer Good Plugins 1.2.1
Release notes for GStreamer Good Plugins 1.2.0
Release notes for GStreamer Good Plugins 1.2.1
The GStreamer team is proud to announce a new feature release
The GStreamer team is proud to announce a new bug-fix release
in the 1.x stable series of the
core of the GStreamer streaming media framework.
......@@ -57,11 +57,26 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 706083 : v4l2src: UVC Allocated buffers wrapped in GstBuffer get orphaned by GstBuffer API
* 707242 : qtmux: streamable and faststart property have no effect
* 683536 : souphttpsrc: Handles long pause (long enough to timeout TCP socket connection) as EOS
* 707933 : matroskademux: Wrong UTF8 detection causes wrong detection of subtitle encoding
* 708501 : osxvideosink: fix segfault releasing the element
* 708622 : rtpjitterbuffer: fix various regressions
* 707975 : qtdemux: Can't handle datetimes before 1970 yet
* 708505 : matroskademux: sends unnecessary gap events
* 708864 : mp4mux: Does not negotiate due to framerate caps not set on peer pad
* 709270 : qtdemux: Does not support mp2v fourcc for MPEG-2 video
* 709352 : jpegdec: Does not require width/height on caps or parsed input
* 709384 : videomixer: Check if the source pad needs reconfiguration and update the source caps in that case
* 709390 : videomixer: Update videoconversion code
* 709423 : v4l2bufferpool: O_CLOEXEC needs _GNU_SOURCE
* 709457 : Do not write out SegmentUID when muxing to WebM