+=== release 1.9.2 ===
+
+2016-09-01 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.9.2
+
+2016-09-01 11:23:33 +0300 Sebastian Dröge <sebastian@centricular.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:
+ po: Update translations
+
+2016-09-01 10:59:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/equalizer/demo.c:
+ * tests/examples/spectrum/demo-audiotest.c:
+ * tests/examples/spectrum/demo-osssrc.c:
+ tests/examples: #define GDK_DISABLE_DEPRECATION_WARNINGS
+ We use gdk_cairo_create() which is deprecated since 3.22.
+
+2016-08-31 05:50:44 +1000 Jan Schmidt <jan@centricular.com>
+
+ * sys/osxvideo/Makefile.am:
+ * sys/osxvideo/cocoawindow.h:
+ * sys/osxvideo/osxvideosink.h:
+ osxvideo: Remove QuickTime references.
+ QuickTime.h is no longer available on OS X 10.12 (Sierra),
+ and both the header and the framework seem unnecessary
+ for compilation - at least as of 10.11 (El Capitan).
+ https://bugzilla.gnome.org/show_bug.cgi?id=770526
+
+2016-08-19 11:11:03 -0700 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * gst/avi/gstavidemux.c:
+ * gst/debugutils/rndbuffersize.c:
+ * gst/flv/gstflvdemux.c:
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/wavparse/gstwavparse.c:
+ Use the new API to post flow ERROR messages on the bus
+ https://bugzilla.gnome.org/show_bug.cgi?id=770158
+
+2016-08-26 21:32:07 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/elements/.gitignore:
+ gitignore: ignore qtdemux, rtph261 and rtpvp9 tests
+
+2016-08-26 21:22:16 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/Makefile.am:
+ tests: use GST_NET_LIBS instead of hardcoded -lgstnet
+ Fixes build in OSX when running 'make check' in gst-uninstalled.
+
+2016-08-26 21:14:47 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: remove a wrong 'const' specifier
+ Fixes "error: duplicate 'const' declaration specifier"
+
+2016-08-26 21:11:59 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ build: silence error about pthread for 'make check' in osx
+ Fixes "clang: error: argument unused during compilation: '-pthread'"
+
+2016-08-26 20:31:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/Makefile.am:
+ vp9enc: Fix build of unit test by letting it link to libgstvideo
+
+2016-08-26 12:06:35 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ Revert "rtpmux: fix PROP_TIMESTAMP_OFFSET range problems"
+ This broke API, so we need a better solution!
+ This reverts commit c7579d31a6e9d788e94b83258309063d0aae481e.
+
+2016-06-08 15:06:28 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtpvp9.c:
+ rtpvp9depay: Support flexible mode
+
+2016-06-06 17:03:36 +0200 Stian Selnes <stian@pexip.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/vp9enc.c:
+ vp9enc: Fix leak of vpx_image_t
+
+2016-05-06 13:33:22 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263pdepay: Don't try to push empty frame
+ If the result of depayloading is an empty frame, just drop it. This is
+ likely the result of a buggy payloader.
+
+2016-05-06 16:06:53 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ rtpmux: fix PROP_TIMESTAMP_OFFSET range problems
+ It could not set the offset for the full guint32 range.
+
+2016-05-06 09:44:42 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: introduce max-streams property
+ To be able to cap the number of allowed streams for one session.
+ This is useful for preventing DoS attacks, where a sender can change
+ SSRC for every buffer, effectively bringing rtpbin to a halt.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770292
+
+2016-03-31 00:10:49 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: reordered packets are very normal, and should not be a warning
+
+2016-02-05 14:19:25 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: degrade g_warning to GST_ERROR
+ So we don't blow up while investigating
+
+2016-02-04 14:16:40 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263pdepay: Fix picture header for non-writable payload
+ Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
+ the payload. In this case the payload modifications will not affect the
+ rtp buffer. So instead of modifying the payload buffer directly we
+ should modify the buffer that actually gets pushed on the adapter.
+
+2015-11-19 11:50:47 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph261depay.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtph261.c:
+ rtph261depay: Fix check of valid payload length
+ Packets with no H.261 payload should be dropped to avoid invalid
+ write/reads.
+
+2015-11-09 10:06:21 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263pay: Fix double free, invalid reads and leak
+
+2014-06-30 15:43:58 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: sanity check RTT before ignoring PLI/FIR
+
+2014-06-30 15:07:45 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: handle sdes messages with non-utf8 more gracefully
+
+2014-06-17 08:52:50 +0200 Stian Selnes <stian.selnes@gmail.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: change log level on bitstream parsing messages
+
+2016-07-07 11:13:18 +0200 Mikhail Fludkov <misha@pexip.com>
+
+ * tests/check/elements/rtprtx.c:
+ tests/rtprtx: refactor the tests to use gstharness
+ The functionality of all the tests was kept exactly the same. Some tests
+ were renamed:
+ test_push_forward_seq -> test_rtxsend_rtxreceive
+ test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
+ test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
+ test_rtxreceive_data_reconstruction was testing that retransmitted
+ buffer produced by rtxsend was correctly transformed to the original
+ buffer by rtxreceive. Now we are checking for this in all the tests
+ where both rtxsend & rtxreceive are involved. That's why the test was
+ removed.
+
+2016-08-25 15:52:36 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Set RTP marker bit
+ Set the RTP marker bit on the last RTP packet of an H.265 access unit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770394
+
+2016-07-26 19:39:58 +0200 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * gst/videofilter/gstvideoflip.c:
+ * gst/videofilter/gstvideoflip.h:
+ videoflip: added GstVideoDirection interface
+ It implements now this interface with its video-direction
+ property. Values are changed to GstVideoOrientationMethod but they have
+ the same value than the originals.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768687
+
+2015-11-06 10:39:16 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ gstrtpsession: refactor duplicate code into a function
+ Less code, easier to read, more consistent.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770293
+
+2016-08-23 17:06:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: fix typo in max-misorder-time property name
+
+2016-08-22 00:05:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: fix printf format compiler warning in debug message
+ On 32-bit x86: gstsplitmuxsink.c:966:31: warning: format ‘%u’ expects
+ argument of type ‘unsigned int’, but argument 9 has type
+ ‘guint64 {aka long long unsigned int}’
+
+2016-08-12 21:12:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * .gitignore:
+ * config.h.meson:
+ * ext/cairo/meson.build:
+ * ext/dv/meson.build:
+ * ext/flac/meson.build:
+ * ext/gdk_pixbuf/meson.build:
+ * ext/jack/meson.build:
+ * ext/jpeg/meson.build:
+ * ext/libpng/meson.build:
+ * ext/meson.build:
+ * ext/pulse/meson.build:
+ * ext/shout2/meson.build:
+ * ext/soup/meson.build:
+ * ext/speex/meson.build:
+ * ext/taglib/meson.build:
+ * ext/vpx/meson.build:
+ * ext/wavpack/meson.build:
+ * gst/alpha/meson.build:
+ * gst/apetag/meson.build:
+ * gst/audiofx/meson.build:
+ * gst/audioparsers/meson.build:
+ * gst/auparse/meson.build:
+ * gst/autodetect/meson.build:
+ * gst/avi/meson.build:
+ * gst/cutter/meson.build:
+ * gst/debugutils/meson.build:
+ * gst/deinterlace/meson.build:
+ * gst/dtmf/meson.build:
+ * gst/effectv/meson.build:
+ * gst/equalizer/meson.build:
+ * gst/flv/meson.build:
+ * gst/flx/meson.build:
+ * gst/goom/meson.build:
+ * gst/goom2k1/meson.build:
+ * gst/icydemux/meson.build:
+ * gst/id3demux/meson.build:
+ * gst/imagefreeze/meson.build:
+ * gst/interleave/meson.build:
+ * gst/isomp4/meson.build:
+ * gst/law/meson.build:
+ * gst/level/meson.build:
+ * gst/matroska/meson.build:
+ * gst/meson.build:
+ * gst/monoscope/meson.build:
+ * gst/multifile/meson.build:
+ * gst/multipart/meson.build:
+ * gst/replaygain/meson.build:
+ * gst/rtp/meson.build:
+ * gst/rtpmanager/meson.build:
+ * gst/rtsp/meson.build:
+ * gst/shapewipe/meson.build:
+ * gst/smpte/meson.build:
+ * gst/spectrum/meson.build:
+ * gst/udp/meson.build:
+ * gst/videobox/meson.build:
+ * gst/videocrop/meson.build:
+ * gst/videofilter/meson.build:
+ * gst/videomixer/meson.build:
+ * gst/wavenc/meson.build:
+ * gst/wavparse/meson.build:
+ * gst/y4m/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ * sys/directsound/meson.build:
+ * sys/meson.build:
+ * sys/v4l2/meson.build:
+ * sys/ximage/meson.build:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ Add support for Meson as alternative/parallel build system
+ https://github.com/mesonbuild/meson
+ With contributions from:
+ Tim-Philipp Müller <tim@centricular.com>
+ Jussi Pakkanen <jpakkane@gmail.com> (original port)
+ Highlights of the features provided are:
+ * Faster builds on Linux (~40-50% faster)
+ * The ability to build with MSVC on Windows
+ * Generate Visual Studio project files
+ * Generate XCode project files
+ * Much faster builds on Windows (on-par with Linux)
+ * Seriously fast configure and building on embedded
+ ... and many more. For more details see:
+ http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
+ http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
+ Building with Meson should work on both Linux and Windows, but may
+ need a few more tweaks on other operating systems.
+
+2016-08-20 16:59:30 +0800 Jie Jiang <jiangjie@nudt.edu.cn>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ Fixed splitmuxsink 32-bit overflow bug
+ Extend the byte tracking counters to 64-bit on
+ all platforms, instead of using gsize, which overflows
+ after 4GB.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770019
+
+2016-08-19 17:18:16 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4: Fix coverity warning
+ If atom_copy_data fails to write anything, return 0
+ CID #1371458
+
+2016-04-09 07:51:03 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ * sys/v4l2/v4l2-utils.c:
+ v4l2: consistently check #ifdef HAVE_GUDEV instead of #if
+ Both work with autotools but they definitely don't mean the same thing, cause
+ problems with other build systems, and are bad form. Existence should always be
+ checked with #ifdef or #if defined.
+
+2016-04-19 10:53:05 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ * sys/directsound/gstdirectsoundsink.h:
+ directsound: port away from old DirectX API
+ D3DX has been deprecated for the last 4 years and latest versions of
+ Windows no longer ship headers for it. This is fine as long as you're
+ building with Cerbero's Wine-based DirectX headers, but sucks if you
+ want to build against the actual Windows SDK.
+ We were just using it to get error strings anyway, so just use the
+ generic error string API.
+
+2016-08-18 12:02:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ Revert "flacparse: Add maximum bitrate tag"
+ This reverts commit c703ab69f526092bb26cce41ca691a896c8383d8.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769392
+
+2016-08-18 09:57:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix unit test by disabling adaptive misorder/dropout calculations
+ Need to set max-misorder-time and max-dropout-time to 0 so the
+ jitterbuffer does not base them on packet rate calculations.
+ If it does, out gap is big enough to be considered a new stream and
+ we wait for a few consecutive packets just to be sure
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2016-08-09 12:55:59 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Add option to split at exactly max-size-time
+ Will try to request a keyframe from the encoder to be sent at the target
+ running time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769664
+
+2016-08-09 20:16:16 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Allow time and bytes to reach their respective thresholds
+ https://bugzilla.gnome.org/show_bug.cgi?id=769664
+
+2016-08-17 09:49:04 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Allow mimetypes with properties as long as they're application/sdp
+ Some servers add properties like charset, e.g.
+ application/sdp; charset=utf8
+ Ideally we should also parse the charset and do conversion of all messages,
+ but that's for a later time.
+
+2016-06-24 16:32:37 +0300 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Added support for writing timecode track
+ https://bugzilla.gnome.org/show_bug.cgi?id=767950
+
+2016-08-11 16:32:21 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: Initialize bytes_sent field.
+ This fixes endpoints not receiving any data intermittently.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769773
+
+2016-08-10 11:45:13 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpstats.c:
+ rtpjitterbuffer: Actually calculate the packet rate for max-dropout and max-misorder calculations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2016-08-10 11:26:17 -0600 Thomas Bluemel <tbluemel@control4.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Don't warn for duplicate packets
+ This is a normal scenario and should not be a warning. This can
+ happen frequently when re-transmits of lost packets are enabled.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762208
+
+2016-08-08 13:49:19 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: Fix typo converting to running time.
+ Use the correct collected timestamp.
+
+2016-08-08 02:53:48 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ Revert "splitmuxsink: Use GstBin async-handling instead of our own."
+ This reverts commit fa008f271a52f82dededc28bd81b020ca7939b47.
+ async-handling in GstBin causes the pipeline to spin at 100%
+ CPU as the top-level pipeline tries to change that state
+ to PLAYING constantly. This is a workaround for a core
+ problem, essentially, but an improvement in this case for now.
+
+2016-08-08 00:56:38 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: Recheck state after unlocking mutex.
+ After dropping the splitmux lock, re-check the state,
+ don't just fall through and sleep unconditionally,
+ as we may have already missed the wakeup.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769514
+
+2016-08-03 03:32:07 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Don't stop and error on EOS flow return
+ Don't immediately halt on EOS flow return from downstream
+ due to out of segment. Let the demuxer handle it and send
+ EOS.
+
+2016-08-04 00:36:28 -0300 Thiago Santos <thiagossantos@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: avoid unref of null buffer
+ The current 'l' pointer will be NULL when the loop
+ is interrupted with a 'break' statement. Need to have
+ it advance to the next list item before interrupting.
+
+2016-08-02 14:01:14 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/wavparse/Makefile.am:
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Add tags for container format and bitrate for uncompressed PCM
+ The PCM bitrate is added to help downstream elements (like uridecodebin)
+ figure out a proper network buffer size
+ https://bugzilla.gnome.org/show_bug.cgi?id=769390
+
+2016-08-01 18:52:26 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: Add maximum bitrate tag
+ https://bugzilla.gnome.org/show_bug.cgi?id=769392
+
+2016-07-28 17:58:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: When receiving a DISCONT buffer that does not point to a sample, remember the offset
+ And don't just reset everything. This makes sure that we can continue to
+ handle data in the following scenario:
+ moov: discont
+ moof: discont
+ mdat: continuous
+ Previously this would fail because the offset would be the accumulated offset
+ from moov and moof at the mdat position, while the buffer offset might be
+ something completely different.
+
+2016-07-25 13:34:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpbvpay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtpilbcpay.c:
+ rtp: Filter with the filter caps in the payloader's getcaps
+
+2016-03-03 11:35:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: include http-status-code in error message details
+ https://bugzilla.gnome.org/show_bug.cgi?id=763038
+
+2016-07-25 18:20:03 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix debug statement signedness.
+ The ts variable is a GstClockTime, don't print it
+ as a GstClockTimeDiff.
+
+2016-07-17 22:41:02 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Handle negative running time
+ Use signed clock times for running time everywhere
+ so that we handle negative running times without
+ going haywire, similar to what queue and multiqueue
+ do these days.
+
+2016-07-18 00:12:55 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Drop lock when sending dummy event
+ When pushing the dummy event into the multiqueue,
+ drop the splitmux lock or else we might deadlock.
+
+2016-06-30 01:56:41 +1000 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Intersect with filter caps in getcaps function.
+ Always intersect with the filter caps in the getcaps function
+ to make sure we return a subset of what was requested.
+ Other payloaders also have this problem and need fixing
+ in future commits.
+
+2016-07-12 17:30:56 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/qtdemux.c:
+ tests: qtdemux: fix element and pad leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=768739
+
+2016-07-12 16:45:36 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/audiofirfilter.c:
+ * tests/check/elements/audioiirfilter.c:
+ * tests/check/elements/rtp-payloading.c:
+ * tests/check/elements/videobox.c:
+ * tests/check/pipelines/effectv.c:
+ tests: fix bus leaks
+ gst_bus_add_signal_watch() takes a ref on the bus which should be
+ released using gst_bus_remove_signal_watch().
+ https://bugzilla.gnome.org/show_bug.cgi?id=768739
+
+2016-07-14 03:07:11 +0800 Ting-Wei Lan <lantw@src.gnome.org>
+
+ * configure.ac:
+ configure: Call AG_GST_PKG_CONFIG_PATH to set GST_PKG_CONFIG_PATH
+ GST_PKG_CONFIG_PATH is used in docs/plugins directory, so
+ AG_GST_PKG_CONFIG_PATH must be called to set it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768787
+
+2016-07-12 07:39:58 +0200 Edward Hervey <edward@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Don't drop final bytes of a range request
+ At the end of a range request, we don't want to return GST_FLOW_EOS otherwise
+ the last bytes we just read will be dropped by basesrc.
+ Instead just return GST_FLOW_OK (which was set just before) and let basesrc
+ handle the fact we are at the end of the segment.
+
+2016-07-11 18:30:18 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2provider: Fix device type detection
+ The type detection would lead to assertion as it would try
+ to create a device without having found any type for it. It
+ also didn't detect MPLANE devices properly.
+
+2016-07-11 18:29:01 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't assert when used by the monitor
+ The monitor sets the object->element object as a GstObject. This
+ works for debug traces, but will assert for ELEMENT_ERROR. This
+ was the only case where that could happen. Add a check for that.
+
+2016-07-11 17:38:00 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Indent very long line
+
+2016-07-12 00:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: At the end of a range request, read another time to finalize the request
+ If we're at the end of a range request, read again to let libsoup
+ finalize the request. This allows to reuse the connection again later,
+ otherwise we would have to cancel the message and close the connection.
+
+2016-07-11 21:13:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f363b32 to f49c55e
+
+2016-07-11 19:57:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Fix keep-alive handling
+ We have to get rid of the message on EOS when the complete stream is read to
+ remember that we successfully finished handling this specific message.
+ Otherwise we will cancel it later and close the connection instead of reusing
+ it at a later time.
+ It might also make sense to reuse connections if a non-200 response is
+ received. As long as there was no connection error, the HTTP connection should
+ be re-usable.
+
+2016-07-11 12:05:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ Also enable V4L2 probe on aarch64 (aka ARM 64bit)
+
+2016-07-11 11:59:19 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/examples/rtp/client-PCMA.c:
+ rtp example: Fix leak
+ Also stop fetching the internal source as this
+ functionality has been broken.
+
+2016-07-08 14:58:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ Enable v4l2 probe on Linux/ARM
+ Most of those have V4L2 drivers these days enabling it make sure that it
+ this code is enabled in major distribution, hence that HW accelerated
+ decoder/encoder can be used on platforms that support it. The probes are
+ slightly increasing the first init of gstreamer library, though the
+ result is cached in the registry for later use.
+
+2016-07-11 09:46:49 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ * tests/check/elements/rtp-payloading.c:
+ rtph265pay: Accept array_completeness=1
+ When parsing NAL unit type in codec_data, check the 6bits of
+ NAL_unit_type only and do not require the array_completeness bit to be
+ 0, since the default and mandatory value of array_completeness is 1 for
+ hvc1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768653
+
+2016-07-10 21:35:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Also copy device_caps in gst_v4l2_dup
+ This fixes regression where M2M error out saying they have no output
+ format (the V4L2 CAPTURE side).
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-10 21:30:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Use correct in6_pktinfo struct instead of in_pktinfo
+ Fixes the build on FreeBSD, which does not have the latter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768623
+
+2016-07-08 17:28:19 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: fix multiplanar capture
+ After switching to using V4L2_CAP_DEVICE_CAPS we lost support for
+ multiplanar device types. After some research, it looks like
+ vcap.capabilities treated the multiplanar flag of output and capture
+ devices equally, but not the new device_caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-08 14:56:30 +0200 Mats Lindestam <matslm@axis.com>
+
+ * gst/multipart/multipartmux.c:
+ * gst/multipart/multipartmux.h:
+ multipartmux: Use PTS and DTS instead of timestamp
+ And pass-through both of them.
+ Based on a patch by Göran Jönsson <goranjn@axis.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=767900
+
+2016-06-30 14:40:40 +0200 Thomas Scheuermann <Thomas.Scheuermann@barco.com>
+
+ * ext/jack/gstjackaudioclient.c:
+ jack: don't wait for callbacks if the jack server shut down
+ Otherwise we'll wait forever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747275
+
+2016-06-23 15:30:19 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Let upstream events go through upstream
+ There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+ Some elements might want to have that information.
+
+2016-06-23 15:22:56 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Let upstream events go through upstream
+ There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+ Some elements might want to have that information.
+
+2016-06-23 15:17:36 +0200 Edward Hervey <edward@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Let upstream events go through upstream
+ There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
+ Some elements might want to have that information.
+ Also remove downstream-only CAPS event handling and minimize code
+
+2016-07-07 23:53:54 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2.c:
+ v4l2: fix v4l2 probe build error
+ A typo in gst_v4l2_probe_and_register() caused a build error when building
+ with --enable-v4l2-probe. Fixing it.
+ gstv4l2.c: In function 'gst_v4l2_probe_and_register':
+ gstv4l2.c:150:25: error: 'struct v4l2_capability' has no member named 'capabilitites'
+ device_caps = vcap.capabilitites;
+
+2016-07-01 22:53:33 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: use gst_caps_intersect_full in negotiate()
+ Instead of reimplementing the GST_CAPS_INTERSECT_FIRST
+ interection mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-02 01:56:07 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2radio.c:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: use opened device caps instead of physical device ones
+ The same physical device can export multiple devices. In
+ this case, the capabilities field now contains a union of
+ all caps available from all exported V4L2 devices alongside
+ a V4L2_CAP_DEVICE_CAPS flag that should be used to decide
+ what capabilities to consider. In our case, we need the
+ ones from the exported device we are using.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768195
+
+2016-07-07 18:24:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Remove suspicious checks for pads being active and linked
+ We should add all pads, no matter if they are linked or active or not at this
+ point. Skipping some that are not will cause different behaviour than with
+ other muxers.
+
+2016-07-07 18:23:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Error out if we start writing data with some pads not having a codec id yet
+ This can only happen if a) upstream somehow gets around the CAPS event failing
+ or b) there never being any CAPS event.
+ The following code assumes that all pads have a codec-id.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768509
+
+2016-07-07 18:14:43 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Consistently use gst_matroska_mux_set_codec_id() for setting the codec id
+
+2016-07-04 09:50:11 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtph265pay/depay: Sync against RFC 7798
+ Handle sprop-vps, sprop-sps and sprop-pps in caps instead of
+ sprop-parameter-sets.
+ rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It
+ handles profile-id, tier-flag and level-id in caps query.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-07-06 09:25:00 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ flvdemux: Push nominal bitrate tags
+ Add per-stream tag lists, which are used to send nominal
+ bitrate tags. When remuxing FLV => FLV, this now passes
+ through the upstream bitrate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768440
+
+2016-07-06 09:24:49 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ flvdemux: Refactor metadata tag handling
+ The FLV header cannot be trusted to indicate video or
+ audio presence, as the comments already mention. Don't
+ delay pushing tags waiting for streams that might never
+ appear.
+ Tags are now pushed immediately after they change:
+ - After parsing an onMetaData script object
+ - After negotiating caps on a pad
+ https://bugzilla.gnome.org/show_bug.cgi?id=768440
+
+2016-07-06 12:44:10 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix AAC codec_data values
+ As seen in the parent switch for object_type_id, the 4 possible values are
+ 0x40, 0x66, 0x67 and 0x68. Fixing the nested switch to match these values.
+ Looks like it was a typo making them decimal instead of hexadecimal.
+ CID 1363328
+
+2016-07-06 13:51:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.1 ===
+
+2016-07-06 13:06:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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 1.9.1
+
+2016-07-06 11:46:26 +0300 Sebastian Dröge <sebastian@centricular.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:
+ Update .po files
+
+2016-07-06 11:22:53 +0300 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix error messages to first convert to doubles before division
+
+2016-07-06 10:18:30 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/da.po:
+ * po/hr.po:
+ * po/pt_BR.po:
+ * po/sk.po:
+ po: Update translations
+
+2016-07-05 21:11:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else
+ There's a small window for a race condition otherwise.
+
+2016-07-04 17:45:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/qtmux.c:
+ qtmux: Use complete AAC caps with codec_data in the tests
+
+2016-07-04 16:58:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Reject raw AAC if no codec_data is found in the caps
+ If necessary, a demuxer will have to invent something here but this is only a
+ problem with non-conformant files anyway.
+
+2016-07-04 16:55:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Invent AAC codec_data if none is present
+ Without, raw AAC can't be handled and we have some information available in
+ the decoder that most likely allows us to decode the stream in one way or
+ another. This is the same code already used by matroskademux for the same
+ reasons, and ffmpeg/vlc play such files just fine too by guesswork.
+
+2016-07-04 14:54:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Reject raw AAC caps without codec_data
+ The resulting file is not going to be playable without guesswork and raw caps
+ should always have codec_data.
+
+2016-05-10 15:48:49 +0200 Edward Hervey <edward@centricular.com>
+
+ qtdemux: Handle upstream GAP in push-mode/time segment
+ This is to handle cases where upstream handles the fragmented streaming in TIME
+ segments and sends us data with gaps within fragments. This would happen when dealing
+ with trick-modes.
+ When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
+ it must obey the following rules:
+ * The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
+ * The buffers containing the first sample after a gap:
+ * MUST start at the beginning of a sample,
+ * MUST have the DISCONT flag set,
+ * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767354
+
+2016-07-01 11:54:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/v4l2-utils.c:
+ v4l2: fix potential double-free of error debug string
+ gst_v4l2_clear_error() doesn't work like g_clear_error(), it
+ doesn't NULLify the pointer, so set freed debug string to NULL
+ so it doesn't get freed again if gst_v4l2_clear_error() is
+ called twice on the error.
+ CID 1362901
+
+2016-07-01 10:05:00 +0000 Brad Lackey <blackey@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't disable UDP protocols on redirecting
+ https://bugzilla.gnome.org/show_bug.cgi?id=768232
+
+2016-07-01 17:28:17 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Push caps only when it was updated
+ Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 caused new caps
+ event per moof without consideration of duplication.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768268
+
+2016-06-30 15:01:46 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: fix invalid memory access
+ 10 bytes was allocated for stream_format but size of "byte-stream" is
+ more. Use g_strdup() instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-29 23:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/shout2/gstshout2.c:
+ shout2: Use a non-timer GstPoll
+ Otherwise set_flushing() will have undefined semantics and nowadays causes a
+ g_critical() to warn about that.
+
+2016-06-19 02:08:25 -0300 Thiago Santos <thiagossantos@gmail.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: dynamically adjust blocksize
+ Update the blocksize depending on how much is obtained from a read
+ of the input stream. This avoids doing too many reads in small chunks
+ when larger amounts of data are available and also prevents using
+ a very large memory area to read a small chunk of data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767833
+
+2016-06-28 16:44:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Windows has no ipi_spec_dst in struct in_pktinfo
+
+2016-06-28 15:15:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: #define __APPLE_USE_RFC_3542 to be able to use IPV6_PKTINFO on OSX/iOS
+
+2016-06-28 15:08:04 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Move #includes around to a) work around broken glibc header and b) Windows
+
+2016-06-28 14:25:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix compilation on Windows and *BSD/OSX
+
+2016-06-23 20:21:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Filter out multicast packets that are not for our multicast address
+ https://bugzilla.gnome.org/show_bug.cgi?id=767980
+
+2016-06-28 10:57:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state
+ If we consider the RTSP state, what can happen is that it is PLAYING but the
+ element already asynchronously tried to PAUSE and it just did not happen yet.
+ We would then override this setting to PAUSED (while the element actually is
+ in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
+ to produce packets while the sinks are all PAUSED, piling up thousands of
+ packets in the rtpjitterbuffer and other elements and finally failing.
+
+2016-06-27 09:20:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Add comment about H263/MPEG4P2 being non-standard for FLV
+ They are however supported by ffmpeg and apparently used out there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=768006
+
+2016-06-24 14:48:53 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Add support for H263 and MPEG4 part2
+ https://bugzilla.gnome.org/show_bug.cgi?id=768006
+
+2016-06-21 17:10:56 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * 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:
+ Update plugins doc
+ This is partly automated using "make update" in docs/plugins, but also
+ required manual merge. Additionally, missing plugins and elements have
+ been added.
+
+2016-06-21 17:51:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/splitmux.c:
+ tests: splitmux: skip tests if theora or ogg plugins are not available
+ https://bugzilla.gnome.org/show_bug.cgi?id=767861
+
+2016-06-21 11:46:13 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From ac2f647 to f363b32
+
+2016-06-21 07:40:42 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ gstrtpj2kpay: use tile bit and tile number to determine if there are multiple tiles in packet
+ Now we don't have to rely on a special value for the tile number.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767817
+
+2016-06-21 09:34:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: fix compiler warning on OS/X
+ gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535
+ https://bugzilla.gnome.org/show_bug.cgi?id=767817
+
+2016-06-21 09:34:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ docs: update
+
+2016-05-16 17:31:58 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/capssetter.c:
+ * tests/check/elements/icydemux.c:
+ * tests/check/elements/jpegenc.c:
+ * tests/check/elements/level.c:
+ * tests/check/elements/multifile.c:
+ * tests/check/elements/qtmux.c:
+ * tests/check/elements/rtprtx.c:
+ * tests/check/elements/udpsrc.c:
+ fix buffer leaks in tests
+ Need to call gst_check_drop_buffers() to release the buffers exchanged
+ during the test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-17 12:52:43 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/interleave.c:
+ interleave: fix message leaks in test
+ Flush the bus when cleaning up so pending messages are destroyed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-17 12:58:06 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/videomixer.c:
+ videomixer: fix event leaks in test
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-05-13 15:12:22 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/deinterleave.c:
+ deinterleave: fix leaks
+ - Flush the bus so messages aren't leaked
+ - Fix pad leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=766561
+
+2016-06-17 15:29:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Deprecated sprop-parameter-set property
+ This is supposed to be either in the codec_data (avc stream format) or inside
+ the stream, and we extract it from there. It should not be set from a
+ property as it's stream specific.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767789
+
+2016-06-17 12:16:32 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: make all srtp encoder properties explicit
+ The Session Data Protocol doesn't allow specifying a cipher for the
+ SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
+ "aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
+ an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
+ cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767799
+
+2016-06-17 19:59:13 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/soup/gstsoup.c:
+ soup: work around frequent deadlocks in GLib type initialisation
+ .. by registering the types from the plugin init function. This
+ seems to help, but we'll see if it's enough (might need similar
+ things elsewhere).
+ https://bugzilla.gnome.org/show_bug.cgi?id=693911
+ https://bugzilla.gnome.org/show_bug.cgi?id=674885
+
+2016-06-17 16:08:08 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: The prores variant is stored in the variant field, not format
+ And the caps in the sink pad template already used variant (only).
+
+2016-06-17 13:00:48 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtph265pay: Remove sprop-parameter-sets property
+ There is no valid use case when this property is needed since the values
+ must be in either codec_data or buffer data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-10 16:17:26 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * docs/plugins/scanobj-build.stamp:
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Read NALU type the same way everywhere
+ Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the
+ same way as in other places.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753760
+
+2016-06-17 13:58:33 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ rtpjitterbuffer: fix RTPJitterBufferMode documentation
+ Documentation lacks '@' before each enum values and there was an extra
+ line after symbol section which confuses GTK-Doc parser.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767788
+
+2016-05-23 10:18:48 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: take the lock when changing stats
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-06-15 11:19:43 +0200 Jürgen Slowack <jurgen.slowack@barco.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265: fix NAL unit type parsing and SPS/PPS/VPS detection
+ Fixes sps/pps/vps insertion via the config-interval property.
+ https://bugzilla.gnome.org//show_bug.cgi?id=767680
+
+2016-06-11 12:16:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/pipelines/simple-launch-lines.c:
+ simple-launch-lines: Use correct JPEG2000 caps
+
+2016-06-10 13:43:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix indentation
+
+2016-06-10 13:42:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix date parsing when there are trailing spaces
+ Fixes parsing of "Thu May 11 15:57:46 2006 ".
+ https://bugzilla.gnome.org/show_bug.cgi?id=767496
+
+2016-05-13 15:08:24 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kcommon.h:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ gstrtpj2k: set sampling field required by RFC
+ This field is now required in the sink caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766236
+
+2016-06-09 09:30:48 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Fix unref assertion failure
+ Fix unref assertion failure
+ https://bugzilla.gnome.org/show_bug.cgi?id=767424
+
+2016-05-14 14:46:17 +0200 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Work with non-TIME segments
+ With non-time segments, it now assumes that the arrival time of packets
+ is not relevant and that only the RTP timestamp matter and it produces
+ an output segment start at running time 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766438
+
+2016-06-07 20:53:34 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: Wait for segment event before checking it
+ The heuristic to choose between packetise or not was changed to use the
+ segment format. The problem is that this change is reading the segment
+ during the caps event handling. The segment event will only be sent
+ after. That prevented the decoder to go in packetize mode, and avoid
+ useless parsing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2016-06-06 17:00:22 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Wait for segment event before checking it
+ The heuristic to choose between packetise or not was change to use the
+ segment format. The problem is that this change is reading the segment
+ during the caps event handling. The segment event will only be sent
+ after. That prevented the decoder to go in packetize mode, and avoid
+ useless parsing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2016-06-07 16:42:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Keep part of the input buffer
+ Instead of completely getting rid of the input buffer, copy
+ the metadata, the flags and the timestamp into an empty buffer.
+ This way the decoder base class can copy that information again
+ to the output buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758424
+
+2016-06-07 16:41:58 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Coding style fixes
+
+2016-06-07 16:09:23 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Coding style fixes
+
+2016-06-07 16:04:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2transform.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Add an error return to _try/_set_format
+ This way one can easily ignore errors. Previously, error were always
+ posted ont he bus.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766172
+
+2016-06-07 16:01:55 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/v4l2-utils.c:
+ * sys/v4l2/v4l2-utils.h:
+ v4l2-util: Introduce GstV4l2Error
+ This is to allow returning an error that can easily be sent as
+ message to the application if the element needs it. Using this
+ also allow ignoring errors.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766172
+
+2016-06-07 12:41:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Avoid decide allocation on active pool
+ v4l2src will renegotiate only if the format have changed. As of now,
+ it's not possible to change the allocationw without resetting the
+ camera. To avoid unwanted side effect, simply keep the old allocation
+ if no renegotiation is taking place. This fixes assertion and possible
+ failures in USERPTR or DMABUF import mode (when using downstream pools).
+ https://bugzilla.gnome.org/show_bug.cgi?id=754042
+
+2016-04-28 13:44:49 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Show state name in debugging
+ Makes it easier to trace what's going on
+
+2016-05-10 15:45:42 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove useless variable
+ That variable is only needed for a debug statement, move it there
+
+2016-05-10 15:10:36 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Add/Fix comments on the various structure variables
+ No variables were added/removed. This was just a good excuse to:
+ * Comment what most variables are used for (and when)
+ * Order them in such a way as to show first the common variables used
+ in all cases, followed by those only used in push-mode
+
+2016-05-10 15:07:40 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove unused structure
+ Let's just remove it, been commented for 7+ years :)
+
+2015-09-02 11:48:29 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: use decoder stop command instead of queueing empty buffers
+ Only if the decoder stop command fails, keep queueing empty buffers to
+ signal end of stream as before.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733864
+
+2014-12-12 14:31:36 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: add gst_v4l2_decoder_cmd helper
+ https://bugzilla.gnome.org/show_bug.cgi?id=733864
+
+2016-06-01 20:28:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Forward segments directly if we are operating in PUSH mode on fragmented streams
+ We shouldn't go through segment activation as we will only have a limited
+ understanding of how the whole stream timeline looks like from the moof. We
+ only know about the current fragment, while upstream knows about the whole
+ stream.
+ This fixes seeking in DASH streams, both for seeks after the current moof and
+ for seeks into the current moof. The former would fail because the moof ends
+ and we can't activate any segment, the latter would cause a segment that stops
+ at the moof end, and no further fragments would be played because we end up
+ being EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-06-06 17:54:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: Use looser caps for upstream
+ When we fixate for upstream, try to not introduce new fields when not
+ needed. This was imported from videoconvert element.
+
+2015-01-28 12:07:58 +0100 Enrico Jorns <ejo@pengutronix.de>
+
+ * sys/v4l2/gstv4l2transform.c:
+ gstv4l2transform: format fixation for preferring passthrough
+ * If outgoing format is unfixated, try to set it to input format.
+ * Call gst_caps_fixate () at end of fixation routine
+ https://bugzilla.gnome.org/show_bug.cgi?id=766719
+
+2016-05-20 12:49:53 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: allow to change pixel aspect ratio
+ Scalers may change width and height independently,
+ allow to change pixel aspect ratio.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766712
+
+2016-05-20 12:32:25 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: fix scaling in case of fixed pixel aspect ratio
+ To change pixel aspect ratio from DAR to PAR, the necessary scaling factor
+ is DAR/PAR, not DAR*PAR.
+ For good measure, add debug output similar to the fixed-width and
+ fixed-height cases.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766711
+
+2016-05-13 16:39:25 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fill colorimetry in gst_v4l2_object_acquire_format
+ Instead of relying on the default colorimetry chosen by
+ gst_video_info_set_format(), set info.colorimetry from the
+ values returned by G_FMT. This allows decoders to propagate
+ their input colorimetry downstream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766383
+
+2016-05-18 10:17:12 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: refactor gst_v4l2_object_get_colorspace to take a v4l2_format parameter
+ Move the extraction of colorimetry parameters from struct v4l2_format and the
+ setting of the identity matrix for RGB formats into the function to avoid code
+ duplication.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766383
+
+2016-05-13 14:58:41 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: use visible size, not coded size, for downstream negotiation filter
+ gst_v4l2_probe_caps() returns the coded size, not the visible size. Subtract
+ the known padding from probed caps with the coded size before using them as
+ filter for caps negotiation with downstream elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766382
+
+2016-05-13 14:45:02 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: use G_SELECTION instead of G_CROP in gst_v4l2_object_acquire_format
+ The gst_v4l2_object_acquire_format() function is used by v4l2videodec to obtain
+ the currently set capture format. Since G_FMT returns the coded size, the
+ visible size needs to be obtained from the compose rectangle in order to
+ negotiate it with downstream elements. The G_CROP call hasn't worked on mem2mem
+ capture queues for a long time. Instead use the G_SELECTION call to obtain the
+ compose rectangle and only fall back to G_CROP for ancient kernels.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766381
+
+2016-01-27 09:57:38 +0100 Andreas Naumann <anaumann@ultratronik.de>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: Use V4L2_BUF_TYPE_VIDEO_OUTPUT_OVERLAY if driver advertises it.
+ On modern kernels, the G/S_FMT ioctls will always fail using
+ V4L2_BUF_TYPE_VIDEO_OVERLAY with VFL_DIR_TX (e.g. real overlay out drivers)
+ since this is not the intented use (rather rx, according to v4l2 API doc).
+ Probably this is why the Video Output Overlay interface was created, so if
+ the driver advertises it we might as well use.
+ For old kernels (pre 2012) the old way might still work so keeping this for
+ compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761165
+
+2016-06-06 18:52:01 +0100 Kieran Bingham <kieran@bingham.xyz>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Use non-deprecated V4L2 type for RGB15
+ Support for the updated V4L2_PIX_FMT_XRGB555 was added in commit
+ 2538fee2fd8fdb74b05f0a511281bc4707e7cc44 however, when setting the format
+ for use in v4l2 ioctls, the old deprecated format is still used. Convert
+ this to the new accepted format type, as the preferred format.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767300
+
+2016-05-04 14:50:32 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: preserve seek flags
+ Without this some flags get lost in streaming mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767194
+
+2016-06-06 10:47:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/Makefile.am:
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ Revert "WIP revert soup"
+ This reverts commit fdac3a7a231f3848665636cf8122f96103b46e3b.
+ Was not supposed to be pushed but a local workaround for
+ https://bugzilla.gnome.org/show_bug.cgi?id=693911#c13
+
+2016-06-03 13:09:35 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: complete warn log with SSRC
+ https://bugzilla.gnome.org/show_bug.cgi?id=767195
+
+2016-05-31 15:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/Makefile.am:
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ WIP revert soup
+
+2016-06-03 13:18:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Unref seek event in any case
+ It would be leaked if no seek handler was currently set.
+
+2016-06-03 10:49:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: Properly set event/message sequence numbers based on the previous seek
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-03 10:36:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: Remember if upstream had a time segment and if not properly create time segments
+ Previously the segment.time was wrong, and the position was not updated
+ correctly, resulting in seeks in PUSH mode with upstream providing a BYTES
+ segment to not work at all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-03 09:54:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Implement SEEKING query so we can actually seek if upstream can't seek in TIME
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 14:19:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: Recalculate the frame offsets at the beginning of each BYTE segment and whenever upstream gives us a timestamp
+ This fixes seeking in DV streams where upstream operates in PUSH mode with a
+ TIME segment (e.g. avidemux). Without this, we would generate wrong durations
+ and timestamps after a seek.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 13:53:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: Pass-through buffer DISCONT flags
+ https://bugzilla.gnome.org/show_bug.cgi?id=767157
+
+2016-06-02 16:16:45 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ rtpvp9depay: Don't assert on flexible mode packets
+ Instead just post a warning on the bus for now.
+
+2016-06-02 15:03:17 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/rtpbin.c:
+ tests: rtpbin: fix caps leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=767156
+
+2016-06-02 15:00:01 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * tests/check/elements/amrparse.c:
+ tests: amrparse: clean up test
+ - use GST_CHECK_MAIN() to reduce boilerplate
+ - unref the input caps using a teardown function to prevent leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=767156
+
+2016-05-20 15:22:35 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Ensure DISCONT flag is properly propagated
+ The output of deinterlace at startup, or when receiving a new DISCONT
+ buffer, should have the DISCONT flag set on the first buffer.
+
+2016-05-31 21:34:04 +0200 Josep Torra <adn770@gmail.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2src: check for valid size on raw video buffers
+ Discard buffers that doesn't contain enough data when dealing
+ with raw video inputs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767086
+
+2016-05-31 17:10:36 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use the demuxer segment instead of a new one for MSS streams
+ Upstream might have told us something about the to be expected segment, so
+ let's use that information instead of coming up with a [0,-1] segment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 17:04:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Only activate segments and send SEGMENT events if we have streams
+ But in that case also remove the pending newsegment event, otherwise we would
+ later send a possibly outdated event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 16:53:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: In PULL mode, nothing is ever going to send us a SEGMENT event
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2016-05-31 16:38:34 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't override TIME segments from upstream that we just saw
+ The point of d8fb7a9c96b108814beeaa0e63f818d4648c7fe9 was to not have any
+ spurious segments stored for later if we do BYTES->TIME conversion, but
+ overriding any TIME segments from upstream does not make any sense.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=763165
+ https://bugzilla.gnome.org/show_bug.cgi?id=767071
+
+2015-07-16 09:48:46 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/multifile/gstmultifilesrc.c:
+ multifilesrc: set position as offset from start-index
+ query position in GST_FORMAT_BUFFER returns
+ offset from start-index rather than index.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752462
+
+2016-05-27 12:49:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/files/Makefile.am:
+ * tests/files/gradient.j2k:
+ tests: add unit test for JPEG-2000 rtp payloader leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=766870
+
+2016-05-25 17:11:13 +0200 Pierre Lamot <pierre.lamot@openwide.fr>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: Fix buffer memory leak
+ Input buffer memory was not unmapped
+ https://bugzilla.gnome.org/show_bug.cgi?id=766870
+
+2016-05-18 12:12:15 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fix caps leak
+ gst_v4l2_object_probe_caps() was taking an extra ref on the returned
+ caps for no reason.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766610
+
+2016-05-22 20:14:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ videocrop mark crop properties as mutable in playing state
+
+2016-05-20 16:47:35 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: fix buffer leak when flushing
+ When early returning in gst_soup_http_src_read_buffer() because the
+ element is FLUSHING, we need to unmap and unref the buffer which was just created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766718
+
+2016-05-20 11:15:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Set seek event seqnum on all SEGMENT events
+ Some were forgotten.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-20 11:12:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavidemux.h:
+ avidemux: Pass through seek event seqnums in all SEGMENT/EOS events and SEGMENT_DONE messages/events
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-20 10:56:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Set seek event seqnum in EOS and SEGMENT_DONE messages/events
+ Also actually store the seqnum in pull mode seeks.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=765935
+
+2016-05-17 13:40:38 +0300 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: fix caps leak
+ The caps returned by gst_pad_get_current_caps() was never unreffed when
+ not early returning.
+ Fix a leak with the elements/deinterlace test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766558
+
+2016-01-25 16:25:51 +0100 Mikhail Fludkov <misha@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: don't act on suspicious BYE RTCP
+ Some endpoints (like Tandberg E20) can send BYE packet containing our
+ internal SSRC. I this case we would detect SSRC collision and get rid
+ of the source at some point. But because we are still sending packets
+ with that SSRC the source will be recreated immediately.
+ This brand new internal source will not have some variables incorrectly
+ set in its state. For example 'seqnum-base` and `clock-rate` values will be
+ -1.
+ The fix is not to act on BYE RTCP if it contains internal or unknown
+ SSRC.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762219
+
+2015-11-15 14:54:28 +0100 Mikhail Fludkov <misha@pexip.com>
+
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Add test for locking of the stats signal
+ Keeping the lock while emitting the stats signal introduces potential
+ deadlock in those situations when the signal callback wants the access
+ to rtpsession's properties which also requre the lock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762216
+
+2016-05-19 15:36:57 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: don't hold object lock whilst pushing out headers
+ matroskademux would take the GST_OBJECT_LOCK in
+ - gst_matroska_demux_push_codec_data_all()
+ - gst_matroska_demux_query()
+ Some parse element such as FLAC checks upstream seekability, and
+ there is some use cases that matroska-demux is linked to a parse element
+ (e.g.,FLAC format) without intermediate elements (e.g., queue).
+ In this case, matroska-demux never returns from _push_codec_data_all()
+ because the parser can return only after it receives the response to
+ the upstream query, but that's not going to happen because it's
+ deadlocked.
+ Elements must not hold the object lock whilst pushing out events
+ or data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766645
+
+2016-05-19 12:43:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Set sent_buffers and streamheader_buffers to NULL after freeing
+ Otherwise we might use an already freed list later and crash or worse.
+
+2016-05-18 18:32:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: fix Since version for new "loop" property
+
+2016-05-16 16:18:37 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/rtsp/gstrtpdec.c:
+ rtpdec: fix clock leak
+ gst_system_clock_obtain() returns a new ref.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766521
+
+2016-05-17 05:33:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: add doc blurb with since marker for new "loop" property
+
+2015-11-13 15:52:35 +0100 Dimitrios Katsaros <patcherwork@gmail.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: add support for png
+ https://bugzilla.gnome.org/show_bug.cgi?id=758059
+
+2016-05-15 22:07:14 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxsrc: Connect to demux signals before activating
+ Fix a race in splitmuxsrc by properly connecting to the
+ demuxer signals we're interested in *before* setting it running.
+
+2016-05-15 13:31:37 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ docs: Update for git master
+
+2016-05-15 12:16:23 +0200 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4gpay.h:
+ rtpmp4gpay: Don't produce timestamps based on byte count
+ The GST_BUFFER_OFFSET of output buffers returned to GstRtpBasePayload
+ should reflect the number of "samples" in the unit of the RTP clock in this
+ buffer. If this is not true, then it shouldn't be set.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761943
+
+2016-05-15 12:24:03 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Fix strcmp usage
+ Just use g_strcmp0 which can handle NULL entries
+
+2016-03-04 10:14:00 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Use audio/x-unaligned-raw instead of audio/x-raw for L16 data
+ Directly setting audio/x-raw caps leads to problems when the delivered
+ data blocks do not align properly at sample boundaries (for example, a
+ data block with 391 bytes). So, instead, set audio/x-unaligned-raw to
+ let a parser be autoplugged.
+ https://bugzilla.gnome.org/show_bug.cgi?id=689460
+
+2016-05-12 11:52:09 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Parsing elst box based on version
+ segment_duration and media_time should be parsed based on version
+ of elst box. Specification defines that an elst box with version 1
+ has uint64 and int64 values for segment_duration and media_time,
+ respectively.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766301
+
+2016-05-14 12:57:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: check if request was cancelled when sending message
+ It might be that the request was aborted by the application and
+ we can return immediatelly
+
+2016-05-14 12:43:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: proxy resolver is on by default
+ Remove from the session creation parameters
+
+2016-05-14 12:15:48 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/Makefile.am:
+ soup: update build to warn about newer deprecated functions
+ We already depend on 2.48
+
+2016-05-14 11:09:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: reduce reading latency by using non-blocking read
+ Non-blocking read will return the amount of data available without
+ blocking to wait for the full requested size.
+ The downside is that now it souphttpsrc needs to have a waiting
+ mechanism in case there is no data available yet to avoid busy
+ looping arond the inputstream.
+
+2016-05-15 12:30:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Take the lock already when reading the other stats, not just for the hash table
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-14 17:04:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/ebml-read.c:
+ matroska: use math-compat.h for NAN define
+
+2016-05-14 23:39:22 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Use GstBin async-handling instead of our own.
+ Set the async-handling property on GstBin to let it manage
+ async-handling instead of the local handling from the previous
+ commit. Works because of #174a5e in core
+
+2016-05-13 10:17:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: refactor to use Soup's sync API
+ Replace the async API with the sync API to remove all the extra mainloop
+ and context handling. Currently it blocks reading until 'blocksize'
+ bytes are available but that can be improved by using:
+ https://developer.gnome.org/gio/unstable/GPollableInputStream.html#g-pollable-input-stream-read-nonblocking
+ https://bugzilla.gnome.org/show_bug.cgi?id=693911
+
+2016-05-14 04:50:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: replace deprecated API
+ Avoid using soup_server_run_async and old get_port() APIs,
+ replace with me soup_server_listen and get the port through the
+ URIs list returned from the server.
+
+2016-05-14 12:34:10 +0200 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Upgrade debug message to error
+ It causes the entire pipeline to fail, it should be easier to find.
+
+2016-05-14 18:32:52 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Hide internal async state changes.
+ When switching fragments, hide the async-start/async-done
+ messages from the parent bin, as otherwise we sometimes (very rarely)
+ hang in PAUSED instead of returning / continuing to PLAYING
+ state.
+
+2016-05-13 21:20:28 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Remove stray carriage-return from debug
+
+2016-05-13 16:43:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ rtp: Ship gstrtpj2kcommon.h file to fix distcheck
+
+2015-04-30 14:43:04 +0200 Jesper Larsen <knorr.jesper@gmail.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Do not write index and header if idx is NULL
+ Fixes criticals with e.g.
+ videotestsrc num-buffers=1 ! identity drop-probability=1.0 ! avimux ! fakesink
+ https://bugzilla.gnome.org/show_bug.cgi?id=748700
+
+2016-05-12 08:43:39 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: manage T tile invalidation bit correctly, update tile id in header correctly.
+ 1. according to RFC, T bit is only set when either the RTP packet only contains the J2K main header, or the packet contains tile parts from multiple tiles. This is now being managed correctly in the code. The second scenario cannot happen with our payloader, since tile headers are always placed in their own RTP packet, and so a packet cannot contain tile parts from multiple tiles.
+ However, I have added code to track if multiple tile parts are included in a single RTP packet, in case in the future we want to put header and data in same packet.
+ 2. Old code would set the tile id to zero for all J2K packets. This is now set correctly to the appropriate tile id.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-12 08:41:51 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: manage fragmented headers correctly
+ J2K main header framentation across multiple RTP packets is now handled correctly
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-11 15:04:26 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kcommon.h:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kdepay.h:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpj2kpay.h:
+ rtpj2k: move common code to shared header, code clean up
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-11 15:01:32 -0400 Aaron Boxer <boxerab@gmail.com>
+
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2k: update documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=745187
+
+2016-05-12 14:43:43 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/auparse/gstauparse.c:
+ * gst/auparse/gstauparse.h:
+ auparse: Fix sticky event misordering warning
+ Make sure that src pad has caps before sending segment event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766359
+
+2016-05-11 09:28:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Don't notify about stats property changes while taking the session lock
+ The signal handlers might want to actually get the value of the stats
+ property, which would take the session lock again and deadlock.
+ This was introduced by 2e960e70750a0cb7e1117d0c09d08597866a29ee.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-03 13:59:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: improve edts segment handling after seeks in push mode
+ Properly handle edts segments for push-based operation seeking.
+ We only support edts that a single segment that has media at the end,
+ being preceeded by any number of gap segments.
+ This also allows the qt segment rate to be respected after seeks
+ https://bugzilla.gnome.org/show_bug.cgi?id=765669
+
+2016-05-03 10:41:06 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: properly activate segment with rate != 1.0
+ Also use the qt rate to identify the position within a qt segment
+ to properly translate playback time to qt media time
+ https://bugzilla.gnome.org/show_bug.cgi?id=765669
+
+2016-05-03 11:45:01 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix stall when receiving already lost packet
+ When a packet arrives that has already been considered lost as part of a
+ large gap the "lost timer" for this will be cancelled. If the remaining
+ packets of this large gap never arrives, there will be missing entries
+ in the queue and the loop function will keep waiting for these packets
+ to arrive and never push another packet, effectively stalling the
+ pipeline.
+ The proposed fix conciders parts of a large gap definitely lost (since
+ they are calculated from latency) and ignores the late arrivals.
+ In practice the issue is rare since large gaps are scheduled immediately,
+ and for the stall to happen the late arrival needs to be processed
+ before this times out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765933
+
+2016-05-05 14:18:21 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Take session lock when creating stats
+ The access to the session hash table must happen while the session lock is
+ taken, otherwise another thread might modify the hash table while we're
+ creating the stats.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766025
+
+2016-05-03 21:17:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: update segment when new duration is found
+ Otherwise the old segment will have a shorter stop time and would
+ cause the stream to end too early.
+
+2016-05-04 11:37:29 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: dismember activate_segment into 2 parts
+ One that updates and push a new segment, the other will move the
+ stream to the new segment starting position
+
+2016-05-04 09:30:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdec.c:
+ * ext/dv/gstdvdemux.c:
+ dv: Use correct pixel-aspect-ratio values
+ The previous ones resulted in odd display aspect ratios and were different
+ from the ones used by e.g. ffmpeg. The new ones now result in display aspect
+ ratios of 4:3 and 16:9.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765946
+
+2015-11-09 17:55:09 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/elements/splitmux.c:
+ tests: add splitmuxsrc test for new "format-location" signal
+ https://bugzilla.gnome.org/show_bug.cgi?id=753625
+
+2015-11-09 17:51:12 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: add a format-location signal that allows bypassing the location property
+ This signal allows a user to directly return a sorted list of
+ files to be joined, so that they don't have to follow the
+ filename pattern that the "location" property expects.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753625
+
+2016-05-04 11:15:20 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix deadlock case when source reaches EOS
+ https://bugzilla.gnome.org/show_bug.cgi?id=765072
+
+2016-05-03 22:59:27 -0700 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: simplify and correct header scanning
+ The wav spec tells that 'fmt' (and 'bext' if present) must come before 'data'.
+ There is no requirement for 'fmt' to be first. We already had a list of chunks
+ to skip, but it is easier to just skip any chunk while seeking for 'fmt'.
+ This fixes reading files generated by ProTools.
+
+2016-04-30 22:15:13 +0900 Hyunjun Ko <zzoon@igalia.com>
+
+ * sys/osxaudio/Makefile.am:
+ * sys/osxaudio/gstosxaudio.c:
+ * sys/osxaudio/gstosxaudiodeviceprovider.c:
+ * sys/osxaudio/gstosxaudiodeviceprovider.h:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosink.h:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxaudiosrc.h:
+ osxaudio: Support audio device provider on osx
+ https://bugzilla.gnome.org/show_bug.cgi?id=753265
+
+2016-05-01 15:09:27 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavimux.c:
+ avimux: set audio header rate according to calculated bps in stop_file
+ ... now that set_fields is no longer called there by
+ e538608b3f90539003de21c1db238f3c9b946e30
+
+2016-04-29 15:04:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Store the segment sequence number in the EOS events and SEGMENT_DONE events/message
+ Also instead of storing it per stream, store it globally in the demuxer. It's
+ the same for each stream anyway.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765806
+
+2016-04-11 10:54:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Always bind to ANY when address is a multicast address and not only on Windows
+ For IPv6 addresses, binding to a multicast group does not work on Linux
+ either. Always bind to ANY and then later join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764679
+
+2016-04-26 17:01:49 +0800 Song Bing <b06498@freescale.com>
+
+ * sys/ximage/ximageutil.c:
+ ximageutil: shouldn't implement transform if don't support it
+ shouldn't implement transform if don't support it. Or gst_buffer_copy_into()
+ will print ERROR log.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765583
+
+2016-04-28 16:24:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Allow MPEG-1 Layer 1 and 2 in addition to 3 in MP4
+ Via the MPEG-4 Part 3 spec we can support the other layers too.
+ Also correct the samples per frame calculation for MP3 if it's MPEG-2 or
+ MPEG-2.5.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765725
+
+2016-04-27 20:46:34 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Update caps for TCP whenever they change
+ We only changed them for UDP so far, which caused the wrong seqnum-base and
+ other information to be passed to rtpjitterbuffer/etc when seeking. This
+ usually wasn't that much of a problem as the code there is robust enough, but
+ every now and then it causes us to drop up to 32756 packets before we
+ continue doing anything meaningful.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 20:33:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Ensure to not take caps with the wrong pt for getting the clock-rate
+ Especially the caps on the pad might be out of date, and the new caps would be
+ provided for the current pt via the request-pt-map signal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 18:27:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't propagate spurious state change returns from internal elements further
+ We handle them inside rtspsrc and override them in all other cases anyway, so
+ do the same for "internal" state changes like PAUSED->PAUSED and
+ PLAYING->PLAYING.
+ This keeps unexpected NO_PREROLL to confuse state changes in GstBin.
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=760532
+ https://bugzilla.gnome.org/show_bug.cgi?id=765689
+
+2016-04-27 14:09:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Don't override maximum audio chunk size with the scale again just before writing it
+ set_fields() should only be called in the beginning, otherwise we will never
+ remember the maximum audio chunk size and write a wrong block align... which
+ then causes wrong timestamps and other problems.
+
+2016-04-27 13:53:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Actually store the largest audio chunk size for the VBR case of MP2/MP3
+ 3ea338ce271e1f6a96d2ed49d4472b091f6f8b7e changed avimux to do that, but it
+ never actually kept track of the max audio chunk for MP3 and MP2. These are
+ knowing the hdr.scale only after parsing the frames instead of at setcaps
+ time.
+
+2016-04-25 15:03:14 +0200 Mats Lindestam <matslm@axis.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: Allow setting "socket-v6" without setting "socket" too
+ https://bugzilla.gnome.org/show_bug.cgi?id=764897
+
+2016-04-22 15:02:16 +0100 Mario Sanchez Prada <mario@endlessm.com>
+
+ * ext/vpx/gstvpxenc.c:
+ vpxenc: Properly handle frames with too low duration
+ When a frame's duration is too low, calling gst_util_uint64_scale()
+ to scale its value can result into it being truncated to zero, which
+ will cause the vpx encoder to return an VPX_CODEC_INVALID_PARAM error
+ when trying to encode.
+ To prevent this from happening, we simply ignore the duration when
+ encoding if it becomes zero after scaling, logging a warning message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765391
+
+2016-04-22 15:48:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: fix description of linear interlacing method
+
+2016-04-21 14:08:19 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/flv/gstflvmux.c:
+ flv: Handle the case where we do not get any CollectData in handle_buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=765320
+
+2016-04-11 22:41:20 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Do not use unreliable framerate
+ timescale/1 is unreliable value for framerate. Due to downstream
+ element usually use framerate generated by qtdemux, let it be omitted
+ until the framerate can be reliably calculated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764733
+
+2016-04-21 12:53:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ Revert "qtdemux: expose streams with first moof for fragmented format"
+ This reverts commit d8bb6687ea251570c331038279a43d448167d6ad.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764733
+
+2016-02-09 17:17:09 +0000 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: support seeking of CENC encrypted streams
+ When playing a stream that has been protected by DASH CENC, playback
+ will fail if a seek is performed. Qtdemux produces the error "stream
+ is protected using cenc, but no cenc protection system information
+ has been found" and playback stops.
+ The problem is that gst_qtdemux_reset() gets called as part of the
+ FLUSH during a seek. This function frees the protection_system_ids
+ array. When gst_qtdemux_configure_protected_caps() is called after the
+ seek has completed, the protection_system_ids array is empty and
+ qtdemux is unable to create the correct output caps for the protected
+ stream.
+ This commit changes it to only free the protection_system_ids on
+ hard resets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761787
+
+2016-04-18 14:33:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: add "retrieve-sender-address" property
+ This allows disabling of sender address retrieval, which might
+ be useful in certain scenarios, like when the socket is connected,
+ or the sender address is not of interest (e.g. when receiving an
+ MPEG-TS stream). Disabling sender address retrieval in those
+ cases can have minor performance advantages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=563323
+
+2015-11-26 13:15:06 +0100 Dimitrios Katsaros <patcherwork@gmail.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Change warning handling to break infinite message loop
+ v4l2src can cause an "infinite message loop" when a base control exposed as a
+ property is not provided by the device. In these cases, if in the warning message
+ handling for the bus, the GST_DEBUG_BIN_TO_DOT_FILE* category of functions are used,
+ the src lookup causes a new warning to be posted on the bus, causing a loop.
+ This patch changes the warning for these controls so they are not posted on the bus.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758703
+
+2016-04-15 10:44:02 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ spitmuxsink: Avoid creating small file at EOS
+ When EOS is reached, the current file get closed and the last
+ GOP in the mq was written in a new file.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765072
+
+2016-04-15 19:59:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ scaletempo: S16 uses S32 temporary buffers, float/double their own type
+ Make sure to allocate not only a S16 buffer for S16 but a twice as big one to
+ hold S32.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765116
+
+2016-04-16 02:17:26 +1000 Jan Schmidt <jan@centricular.com>
+
+ * ext/pulse/pulsesink.c:
+ Revert "pulsesink: uncork if needed upon commit"
+ This reverts commit 0dd46accf6d282ff07065852bd91c85c78af3394.
+ With some audiosinks, starting the ringbuffer on the first commit
+ causes audio glitches at startup by starting to output segments
+ from the ringbuffer before it has been filled / fully prerolled. This
+ doesn't usually happen with pulsesink because we map the pulseaudio
+ ringbuffer directly, but we should keep things consistent with
+ other sinks with regards to startup latency, plus it gives more
+ headway to avoid glitching, should the initial 2nd segment take
+ more than 10ms to generate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=657076
+
+2016-04-15 00:46:56 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: add srtp rollover counters from mikey crypto sessions
+ The server can send multiple crypto sessions, one for each SSRC with its
+ own rollover counter. We parse this information and pass it to the SRTP
+ decoder via the "request-key" signal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730540
+
+2016-04-15 14:35:07 +0000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix debug output when resyncing
+ Don't output the pointer value of the time() function as a timestamp
+ by using the correct variable.
+ Fixes build on Raspberry Pi 3.
+
+2016-04-15 11:36:36 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: If no proxy is set by properties, use the default libsoup proxy resolver
+ That is, use whatever system settings there might exist. This is the same
+ behaviour we use in the HTTP source.
+
+2016-04-14 10:01:28 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to ac2f647
+
+2016-04-13 18:45:07 +0100 Damian Ziobro <damian@xmementoit.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Add max_files_number property
+ https://bugzilla.gnome.org/show_bug.cgi?id=744612
+
+2016-04-13 10:57:03 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: drop reference to videomixer 2
+ Fix a small grammar mistake on "overlayed" while at it.
+
+2016-04-13 09:57:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/ximage/ximageutil.c:
+ ximage: Initialize all fields in the meta explicitly
+ The meta is not allocated with all fields initialized to zeroes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764902
+
+2016-04-12 09:41:00 +0000 Paolo Pettinato <ppettina@cisco.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Forward sticky events on buffer lists too, not only on buffers
+ https://bugzilla.gnome.org/show_bug.cgi?id=764933
+
+2016-04-12 15:01:28 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Drain the field history if the caps are changing
+ Otherwise we will use fields from the old caps with everything set up for the
+ new caps, causing crashes and worse.
+ Also don't do anything if the same caps are set twice.
+
+2016-04-12 15:00:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Instead of confusing crashes later, just error out immediately if mapping a video frame fails
+ This probably still crashes but at least we get some hint about what goes
+ wrong instead of random behaviour later.
+
+2016-04-12 11:38:51 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: check stream is available in PIFF parser
+ qtdemux->streams is an array, it will never evaluate to true when comparing
+ to NULL. Instead we want to check the number of streams to make sure the
+ stream is available.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753614
+ CID 1358389
+
+2016-04-12 11:37:36 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "qtdemux: redundant check in PIFF parser"
+ This reverts commit 41e10524f3babdd92aac8c8c9d5b9cdf184c2d4e.
+
+2016-04-12 11:05:50 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: redundant check in PIFF parser
+ qtdemux->streams is an array of size GST_QTDEMUX_MAX_STREAMS, it will never
+ evaluate to true when comparing to NULL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753614
+ CID 1358389
+
+2016-04-12 11:56:08 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: avoid leaking GValues
+ unset the GValue if we don't use it any more to avoid leaks.
+
+2016-04-12 10:15:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix rtp_jitter_buffer_get_ts_diff() fill level calculation
+ The head of the queue is the oldest packet (as in lowest seqnum), the tail is
+ the newest packet. To calculate the fill level, we should calculate tail-head
+ while considering wraparounds. Not the other way around.
+ Other code is already doing this in the correct order.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764889
+
+2016-04-11 10:44:56 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/Makefile.am:
+ rtpmanager: It's GST_LIBS, not GST_LIBS_LIBS
+
+2016-04-11 08:33:17 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix parsing segment duration of empty edit list box
+ For empty edit list, segment-duration in edit list box should not be
+ used for segment event.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764870
+
+2016-04-08 13:05:57 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: make timecodescale configurable
+ In some use cases the default timecodescale will produce blocks with the same timestamp
+ https://bugzilla.gnome.org/show_bug.cgi?id=764769
+
+2016-04-07 13:01:52 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jiterbuffer: Move assertion to the right location
+ We shouldn't have "late" lost timers at that point
+
+2016-03-02 14:25:24 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Speed up lost timeout handling
+ When downstream blocks, "lost" timers are created to notify the
+ outgoing thread that packets are lost.
+ The problem is that for high packet-rate streams, we might end up with
+ a big list of lost timeouts (had a use-case with ~1000...).
+ The problem isn't so much the amount of lost timeouts to handle, but
+ rather the way they were handled. All timers would first be iterated,
+ then the one selected would be handled ... to re-iterate the list again.
+ All of this is being done while the jbuf lock is taken, which in some use-cases
+ would return in holding that lock for 10s... blocking any buffers from
+ being accepted in input... which would then arrive late ... which would
+ create plenty of lost timers ... which would cause the same issue.
+ In order to avoid that situation, handle the lost timers immediately when
+ iterating the list of pending timers. This modifies the complexity from
+ a quadratic to a linear complexity.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-03-02 14:23:01 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Don't create lost events if we don't need them
+ When "do-lost" is set to FALSE we don't use/send the lost events.
+ In that case, don't create them to start with :)
+ https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-03-02 13:57:07 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Add tracing of lock usage
+ Helps with debugging lock usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=762988
+
+2016-02-10 19:56:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2: Don't leak v4l2 objects and props on probe errors
+
+2016-04-04 17:42:03 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: add unit test for jpeg depayloader packet loss handling
+ Make sure it always outputs something that looks like a valid
+ JPEG frame, ie. starts with an SOI marker and ends with an EOI
+ marker.
+
+2016-03-15 03:25:26 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtp/gstrtpjpegdepay.c:
+ rtpjpegdepay: Don't send invalid frames downstream after packet loss or a DISCONT
+ After clearing the adapter due to a DISCONT, as might happen when some packet(s)
+ have been lost, the depayloader was pushing data into the adapter (which had no
+ header due to the clear), creating a headerless frame out of it, and sending it
+ downstream. The downstream decoder would then usually ignore it; unless there
+ were lots of DISCONTs from the jitterbuffer in which case the decoder would reach
+ its max_errors limit and throw an element error. Now we just discard that data.
+ It is probaby not worth trying to salvage this data because non-progressive
+ jpeg does not degrade gracefully and makes the video unwatchable even with
+ low packet loss such as 3-5%.
+
+2016-01-05 16:15:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtpjitterbuffer: Add RFC7273 media clock handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=762259
+
+2015-07-10 09:44:15 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: PIFF box detection and parsing support
+ The PIFF data is stored in a custom UUID box which is parsed and the
+ crypto_info of the element is updated accordingly. This allows
+ downstream decryptors to process and decrypt the protected content.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753614
+
+2016-04-01 12:15:05 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtpvorbisdepay.c:
+ rtpvorbisdepay: remove dead code
+ payload_buffer hasn't been assigned a value before the jumps to
+ switch_failed or packet_short. So the value must be NULL. No need
+ to unmap and unref.
+ CID #1316476
+
+2016-03-31 14:57:20 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: fix leak
+ Free memory of current macroblock once it isn't needed so it isn't leaked
+ by the call of the gst_rtp_h263_pay_B_mbfinder function.
+ if (!(mac = gst_rtp_h263_pay_B_mbfinder (context, gob, mac, mb))) {
+ CID 1212156
+
+2016-03-31 02:15:04 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: Handle a hang draining out at EOS
+ Make sure that all data is drained out when the reference pad
+ goes EOS. Fixes a problem where data that arrives on other
+ pads after the reference pad finishes can stall forever and
+ never pass EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763711
+
+2016-03-18 15:45:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix occasional deadlock when ending file with subtitle
+ Deadlock occurs when splitting files if one stream received no buffer during
+ the first GOP of the next file. That can happen in that scenario for example:
+ 1) The first GOP of video is collected, it has a duration of 10s.
+ max_in_running_time is set to 10s.
+ 2) Other streams catchup and we receive the first subtitle buffer at ts=0 and
+ has a duration of 1min.
+ 3) We receive the 2nd subtitle buffer with a ts=1min. in_running_time is set to
+ 1min. That buffer is blocked in handle_mq_input() because
+ max_in_running_time is still 10s.
+ 4) Since all in_running_time are now > 10s, max_out_running_time is now set to
+ 10s. That first GOP gets recorded into the file. The muxer pop buffers out
+ of the mq, when it tries to pop a 2nd subtitle buffer it blocks because the
+ GstDataQueue is empty.
+ 5) A 2nd GOP of video is collected and has a duration of 10s as well.
+ max_in_running_time is now 20s. Since subtitle's in_running_time is already
+ 1min, that GOP is already complete.
+ 6) But let's say we overran the max file size, we thus set state to
+ SPLITMUX_STATE_ENDING_FILE now. As soon as a buffer with ts > 10s (end of
+ previous GOP) arrives in handle_mq_output(), EOS event is sent downstream
+ instead. But since the subtitle queue is empty, that's never going to
+ happen. Pipeline is now deadlocked.
+ To fix this situation we have to:
+ - Send a dummy event through the queue to wakeup output thread.
+ - Update out_running_time to at least max_out_running_time so it sends EOS.
+ - Respect time order, so we set out_running_tim=max_in_running_time because
+ that's bigger than previous buffer and smaller than next.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763711
+
+2015-11-17 18:17:35 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Add new signal 'on-app-rtcp'
+ Similar to the 'on-feedback-rtcp' signal, but emitted for RTCP APP
+ packets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762217
+
+2016-03-24 15:57:11 +0900 Minjae Kim <nate.kim@lge.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpmanager: Set to initial value for 'ntpns' in get_current_times()
+ Initialize "ntpns" variable to -1 as the OE compiler for some reason doesn't
+ realize that the variable is set in all code paths.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764119
+
+2016-01-31 11:08:38 +1100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ rtpjpegpay: Allow different quantization tables for components 2 and 3
+ RFC 2435 mentions in section 4.1 that U/V use table number 1, but this seems
+ just like an example. Some encoders are not following that and there seems to
+ be no reason to reject their streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761345
+
+2016-03-24 19:23:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: Use threads on multi-core systems
+ This is a redo of commit b848c1b6ffd1e508228820a013f94fb445e4777f. The
+ code was lost when the elements where ported to use a baseclass.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764169
+
+2016-02-29 23:40:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * tests/check/elements/splitmux.c:
+ splitmuxsink: only try to create internal sink if it doesn't exist
+ This allows splitmuxsink to be reused after being put to NULL.
+ Test included
+ https://bugzilla.gnome.org/show_bug.cgi?id=762893
+
+2015-10-01 13:41:23 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: probe all colorspace supported by device
+ A device can support more than one colorspace for a given image
+ dimension and pixel format. So we have to probe all the supported
+ colorspace and not only rely on the default one. Otherwise we could end
+ up with negotiation failure if the caps colorimetry field don't match
+ the v4l2 device default one even if the v4l2 could support such
+ colorimetry.
+ This patch enable probing if colorspace for both capture and output
+ device. It really makes sense for output device since the colorspace
+ shall be set by the application and a little less for capture device
+ which, at the moment, shall provide the colorspace; ie: the v4l2
+ specification seems to not take into account the fact that a capture
+ device could do colorspace conversion.
+ As a side effet, probing takes some times and so sligthly delay v4l2
+ initialization. Note that this patch only probe colorspace and not all
+ colorspace, matrix, transfer and range combination to avoid taking too
+ much time, especially with low-speed devices as full probing do 1782
+ ioctl.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755937
+
+2016-03-24 16:21:56 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/flvdemux.c:
+ check: Fix indentation
+
+2016-03-24 16:20:39 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/flvdemux.c:
+ tests: Remove unused variables
+
+2016-03-16 20:26:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: Return the current caps on the srcpads on caps queries
+ It's not like we could accept any other caps here. The caps are decided by the
+ upstream caps event.
+ Also keep the filter order intact when filtering the results against the
+ filter caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763326
+
+2016-03-24 15:14:23 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix qtdemux memory leak in src_convert function
+ If we don't find the index of the sample correctly in src_convert function,
+ we have to unref about the qtdemux before returning value.
+ So, I have modify it about instead pass qtdemux as a parameter into
+ src_convert function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763973
+
+2016-03-22 13:15:20 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add check condition for fail case in get_duration function
+ Currently, get_duration function always return the TRUE even though
+ it can't be set duration correctly. So, we need to add the else condition
+ about the fail case. Also, we already set the GST_CLOCK_TIME_NONE
+ in this function. So I have modify it which is related code in some
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763968
+
+2016-03-21 10:11:23 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Modify data type of duration in handle_src_query function
+ Data type of duration need to modify from guint64 to GstClockTime
+ for consistency in handle_src_query function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763965
+
+2016-03-18 14:40:58 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Added unit tests for field=auto
+ https://bugzilla.gnome.org/show_bug.cgi?id=763869
+
+2016-03-17 21:21:02 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Added "auto" fields mode
+ The "auto" fields mode will detect the upstream and downstream framerates and
+ will decide to deinterlace all or only top fields.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763869
+
+2016-03-16 20:17:55 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ * tests/check/elements/flvdemux.c:
+ flvdemux: don't emit pad-added until caps are ready
+ In other words, gst_pad_get_current_caps should never return NULL
+ in a pad-added callback from the demuxer.
+ Added tests for the two special cases with AAC and H.264 where this
+ would happen every time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763780
+
+2016-03-04 10:30:12 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/aalib/gstaasink.c:
+ * ext/cairo/gstcairooverlay.c:
+ * ext/dv/gstdvdec.c:
+ * ext/dv/gstdvdemux.c:
+ * ext/flac/gstflacdec.c:
+ * ext/flac/gstflacenc.c:
+ * ext/flac/gstflactag.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufsink.c:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/jpeg/gstjpegdec.c:
+ * ext/jpeg/gstjpegenc.c:
+ * ext/jpeg/gstsmokedec.c:
+ * ext/jpeg/gstsmokeenc.c:
+ * ext/libcaca/gstcacasink.c:
+ * ext/libpng/gstpngdec.c:
+ * ext/libpng/gstpngenc.c:
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesrc.c:
+ * ext/raw1394/gstdv1394src.c:
+ * ext/raw1394/gsthdv1394src.c:
+ * ext/shout2/gstshout2.c:
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/speex/gstspeexdec.c:
+ * ext/speex/gstspeexenc.c:
+ * ext/taglib/gstapev2mux.cc:
+ * ext/taglib/gstid3v2mux.cc:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * ext/wavpack/gstwavpackenc.c:
+ * gst/alpha/gstalpha.c:
+ * gst/alpha/gstalphacolor.c:
+ * gst/apetag/gstapedemux.c:
+ * gst/audiofx/audiopanorama.c:
+ * gst/audiofx/gstscaletempo.c:
+ * 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:
+ * gst/auparse/gstauparse.c:
+ * gst/autodetect/gstautoaudiosink.c:
+ * gst/autodetect/gstautoaudiosrc.c:
+ * gst/autodetect/gstautovideosink.c:
+ * gst/autodetect/gstautovideosrc.c:
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavimux.c:
+ * gst/avi/gstavisubtitle.c:
+ * gst/cutter/gstcutter.c:
+ * gst/debugutils/breakmydata.c:
+ * gst/debugutils/cpureport.c:
+ * gst/debugutils/gstcapsdebug.c:
+ * gst/debugutils/gstcapssetter.c:
+ * gst/debugutils/gstnavigationtest.c:
+ * gst/debugutils/gstnavseek.c:
+ * gst/debugutils/gstpushfilesrc.c:
+ * gst/debugutils/gsttaginject.c:
+ * gst/debugutils/progressreport.c:
+ * gst/debugutils/rndbuffersize.c:
+ * gst/debugutils/testplugin.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/dtmf/gstdtmfsrc.c:
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gstedge.c:
+ * gst/effectv/gstop.c:
+ * gst/effectv/gstquark.c:
+ * gst/effectv/gstradioac.c:
+ * gst/effectv/gstrev.c:
+ * gst/effectv/gstripple.c:
+ * gst/effectv/gstshagadelic.c:
+ * gst/effectv/gststreak.c:
+ * gst/effectv/gstvertigo.c:
+ * gst/effectv/gstwarp.c:
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvmux.c:
+ * gst/goom/gstgoom.c:
+ * gst/goom2k1/gstgoom.c:
+ * gst/icydemux/gsticydemux.c:
+ * gst/id3demux/gstid3demux.c:
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/interleave/deinterleave.c:
+ * gst/interleave/interleave.c:
+ * gst/isomp4/gstrtpxqtdepay.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/law/alaw-decode.c:
+ * gst/law/alaw-encode.c:
+ * gst/law/mulaw-decode.c:
+ * gst/law/mulaw-encode.c:
+ * gst/level/gstlevel.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/webm-mux.c:
+ * gst/monoscope/gstmonoscope.c:
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesrc.c:
+ * gst/multifile/gstsplitfilesrc.c:
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/multipart/multipartdemux.c:
+ * gst/multipart/multipartmux.c:
+ * gst/replaygain/gstrganalysis.c:
+ * gst/replaygain/gstrglimiter.c:
+ * gst/replaygain/gstrgvolume.c:
+ * gst/rtp/gstasteriskh263.c:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpL24pay.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/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726pay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpilbcpay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4apay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmparobustdepay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtppcmupay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpsirenpay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpstreamdepay.c:
+ * gst/rtp/gstrtpstreampay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvp9depay.c:
+ * gst/rtp/gstrtpvp9pay.c:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawpay.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpdtmfmux.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpptdemux.c:
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/rtsp/gstrtpdec.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/shapewipe/gstshapewipe.c:
+ * gst/smpte/gstsmpte.c:
+ * gst/smpte/gstsmptealpha.c:
+ * gst/udp/gstdynudpsink.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ * gst/videobox/gstvideobox.c:
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videofilter/gstgamma.c:
+ * gst/videofilter/gstvideobalance.c:
+ * gst/videofilter/gstvideoflip.c:
+ * gst/videofilter/gstvideomedian.c:
+ * gst/videomixer/videomixer2.c:
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavparse/gstwavparse.c:
+ * gst/y4m/gsty4mencode.c:
+ * sys/directsound/gstdirectsoundsink.c:
+ * sys/oss/gstosssink.c:
+ * sys/oss/gstosssrc.c:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxvideo/osxvideosink.m:
+ * sys/sunaudio/gstsunaudiosink.c:
+ * sys/sunaudio/gstsunaudiosrc.c:
+ * sys/waveform/gstwaveformsink.c:
+ * sys/ximage/gstximagesrc.c:
+ * tests/check/elements/autodetect.c:
+ * tests/check/elements/qtmux.c:
+ good: use new gst_element_class_add_static_pad_template()
+ https://bugzilla.gnome.org/show_bug.cgi?id=763076
+
+2016-03-04 09:42:44 +0100 David Buchmann <david.buchmann@gmail.com>
+
+ * tests/check/elements/flvmux.c:
+ flvmux: Test to verify flvmux handles DTS with GST_CLOCK_TIME NONE
+ https://bugzilla.gnome.org/show_bug.cgi?id=762207
+
+2015-11-04 14:51:19 +0900 Jihae Yi <jihae.yi@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: avoid potentially overflowing expression
+ https://bugzilla.gnome.org/show_bug.cgi?id=757569
+
+2016-03-22 10:43:45 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add the function to get channels and sample rate for AAC
+ Add aac_get_channels and sample_rate function to get these value for
+ AAC.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749110
+
+2016-03-24 13:33:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.8.0 ===
+
+2016-03-24 12:27:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-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 1.8.0
+
+2016-03-24 12:02:59 +0200 Sebastian Dröge <sebastian@centricular.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:
+ Update .po files
+
+2016-03-16 20:18:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: Use GstIterator for iterating all pads instead of manually iterating them while holding the object lock all the time
+ Doing queries while holding the object lock is a bit dangerous, and in this
+ case causes deadlocks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763326
+
+2016-03-17 20:53:27 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Fix typo to not change the input caps but our filtered caps
+ Changing the input caps and not using them anymore afterwards is useless, and
+ it breaks negotiation in pipelines like:
+ gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" !
+ deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" !
+ fakesink
+
+=== release 1.7.91 ===
+
+2016-03-15 12:04:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-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 1.7.91
+
+2016-03-15 11:53:37 +0200 Sebastian Dröge <sebastian@centricular.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/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/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
+
+2016-03-15 11:41:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/hu.po:
+ * po/sr.po:
+ po: Update translations
+
+2016-03-15 03:26:14 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/rtpsource.c:
+ rtpmanager: Some comment and documentation clarifications/fixes
+
+2016-03-13 10:33:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ Revert "flacparse: push tags in pre_push_frame"
+ This reverts commit 4065fcb80a49924b70f0c8fc159dec0ff47943a1.
+ flacparse should not push tags by itself, the base class is going to do that
+ while properly merging in upstream tags. It just didn't because of a bug in
+ the base class, which was hidden by this commit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763553
+
+2016-02-25 05:17:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtp/dboolhuff.c:
+ * gst/rtp/dboolhuff.h:
+ * gst/rtp/gstrtpsbcpay.c:
+ win32: Don't use __attribute__ on MSVC
+ Use MSVC-equivalents for alignment and packing compiler directives when building
+ on MSVC
+
+2016-02-25 05:16:42 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/matroska/ebml-read.c:
+ win32: Don't try to include xmath.h on newer Visual Studio
+
+2016-02-25 05:16:09 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ * gst/goom/gstgoom.c:
+ * gst/goom2k1/gstgoom.c:
+ * gst/monoscope/gstmonoscope.c:
+ gst Factor out endian-order RGB formats
+ MSVC seems to ignore preprocessor conditionals inside static pad
+ template macros.
+
+2016-03-08 17:37:17 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ dirctsoundsink: Setting volume should not unmute
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:57:24 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ dirctsoundsink: Fix volume reset on unmute
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:03:55 +0100 Alban Bedel <alban.bedel@avionic-design.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fix capture with bayer formats other than bggr
+ gst_v4l2_object_get_caps_info() always return V4L2_PIX_FMT_SBGGR8
+ for all bayer formats. This is obviously broken if the device use
+ another ordering. Fix this by properly reading the format parameter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763318
+
+2016-03-07 10:28:06 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: reset pending segment if we are already pushing one
+ When upstream is running in bytes in push-mode, qtdemux will
+ convert seeks from time to bytes and send it upstream. Upstream
+ element will perform a byte seek and send a byte segment to qtdemux
+ that will convert it to time and push it downstream.
+ There is, however, the pending_segment variable that stores a new
+ segment event to be pushed before the next data. When handling seeks
+ as mentioned above this variable was being ignored and, if it contained
+ some segment event, it would override the one resulting from the seek.
+ This would restore a previous segment and would cause the seek segment
+ to be discarded downstream.
+ This patch fixes this issue by unrefing any pending segment as the
+ seek from upstream should contain the latest one that should be
+ used, as requested by the application.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763165
+
+2016-03-07 10:27:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: run gst-indent
+ Otherwise commits will fail with our indent check hook
+
+2016-03-04 15:09:45 +0100 Josep Torra <n770galaxy@gmail.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: fix colorimetry for NV12
+ Replicate V4L2_MAP_QUANTIZATION_DEFAULT macro behavior.
+ At #v4l it was described that documentation might be wrong and that
+ we should trust this macro instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762529
+
+2016-03-04 14:07:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix multicast group joining with provided sockets on Windows
+ On Windows the socket will be bound to ANY instead of the multicast group,
+ as binding to a multicast group does not work. Which would mean that we
+ override src->addr to become ANY and won't automatically join a multicast
+ group anymore on Windows.
+ On Linux we would automatically join a multicast group, keep it consistent.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763093
+
+2016-03-02 13:13:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases"
+ This reverts commit a7fb7b53592d87f7983544debb74d364fc3257ad.
+ The mutex is taken by the caller, we should keep it locked when returning so
+ the caller can unlock it again.
+
+2016-03-01 15:01:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: push tags in pre_push_frame
+ Push a tag event before pre-roll if we have tags.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762660
+
=== release 1.7.90 ===
-2016-03-01 Sebastian Dröge <slomo@coaxion.net>
+2016-03-01 18:15:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.7.90
+ * 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 1.7.90
+
+2016-03-01 17:03:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/ca.po:
+ * po/da.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.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/or.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/tr.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
2016-03-01 16:53:27 +0200 Sebastian Dröge <sebastian@centricular.com>
* ext/pulse/Makefile.am:
* ext/pulse/plugin.c:
- * ext/pulse/pulsedevicemonitor.c:
- * ext/pulse/pulsedevicemonitor.h:
* ext/pulse/pulsedeviceprovider.c:
* ext/pulse/pulsedeviceprovider.h:
* sys/v4l2/Makefile.am:
* sys/v4l2/gstv4l2.c:
- * sys/v4l2/gstv4l2devicemonitor.c:
- * sys/v4l2/gstv4l2devicemonitor.h:
* sys/v4l2/gstv4l2deviceprovider.c:
* sys/v4l2/gstv4l2deviceprovider.h:
Rename GstDeviceMonitor to GstDeviceProvider
2013-09-10 16:44:53 -0600 Seán de Búrca <leftmostcat@gmail.com>
- * gst/videomixer/blendorc-dist.c:
- * gst/videomixer/blendorc-dist.h:
* gst/videomixer/videomixerorc-dist.c:
* gst/videomixer/videomixerorc-dist.h:
videomixer: fix build if orc is not installed
* sys/osxaudio/gstosxaudiosrc.h:
* sys/osxaudio/gstosxcoreaudiocommon.c:
* sys/osxaudio/gstosxcoreaudiocommon.h:
- * sys/osxaudio/gstosxringbuffer.c:
- * sys/osxaudio/gstosxringbuffer.h:
osxaudio: port to v.1.0
2013-04-16 19:29:48 -0400 Olivier Crête <olivier.crete@collabora.com>
2012-09-16 15:33:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
- * ext/vp8/GstVP8Enc.prs:
- * ext/vp8/Makefile.am:
- * ext/vp8/gstvp8dec.c:
- * ext/vp8/gstvp8dec.h:
- * ext/vp8/gstvp8enc.c:
- * ext/vp8/gstvp8enc.h:
- * ext/vp8/gstvp8utils.c:
- * ext/vp8/gstvp8utils.h:
- * ext/vp8/plugin.c:
* ext/vpx/GstVP8Enc.prs:
* ext/vpx/Makefile.am:
* ext/vpx/gstvp8dec.c:
* configure.ac:
* ext/gdk_pixbuf/Makefile.am:
- * ext/gdk_pixbuf/gstgdkpixbuf.c:
- * ext/gdk_pixbuf/gstgdkpixbuf.h:
* ext/gdk_pixbuf/gstgdkpixbufdec.c:
* ext/gdk_pixbuf/gstgdkpixbufdec.h:
* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
- * docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpmanager.c:
* ext/soup/gstsoup.c:
* ext/soup/gstsouphttpclientsink.c:
* ext/soup/gstsouphttpclientsink.h:
- * ext/soup/gstsouphttpsink.c:
- * ext/soup/gstsouphttpsink.h:
soup: rename souphttpsink to souphttpclientsink
To avoid confusion, and because we might want a server
sink at some point too.
* sys/v4l2/gstv4l2sink.c:
* sys/v4l2/gstv4l2videooverlay.c:
* sys/v4l2/gstv4l2videooverlay.h:
- * sys/v4l2/gstv4l2xoverlay.c:
- * sys/v4l2/gstv4l2xoverlay.h:
v4l2: update for GstXOverlay => GstVideoOverlay rename
2011-08-07 12:23:26 +0200 Sjoerd Simons <sjoerd.simons@collabora.co.uk>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/inspect/plugin-isomp4.xml:
- * docs/plugins/inspect/plugin-quicktime.xml:
* gst-plugins-good.spec.in:
* gst/isomp4/LEGAL:
* gst/isomp4/Makefile.am:
* gst/isomp4/qtdemux_types.c:
* gst/isomp4/qtdemux_types.h:
* gst/isomp4/qtpalette.h:
- * gst/quicktime/LEGAL:
- * gst/quicktime/Makefile.am:
- * gst/quicktime/atoms.c:
- * gst/quicktime/atoms.h:
- * gst/quicktime/atomsrecovery.c:
- * gst/quicktime/atomsrecovery.h:
- * gst/quicktime/descriptors.c:
- * gst/quicktime/descriptors.h:
- * gst/quicktime/fourcc.h:
- * gst/quicktime/ftypcc.h:
- * gst/quicktime/gstqtmoovrecover.c:
- * gst/quicktime/gstqtmoovrecover.h:
- * gst/quicktime/gstqtmux-doc.c:
- * gst/quicktime/gstqtmux-doc.h:
- * gst/quicktime/gstqtmux.c:
- * gst/quicktime/gstqtmux.h:
- * gst/quicktime/gstqtmuxmap.c:
- * gst/quicktime/gstqtmuxmap.h:
- * gst/quicktime/gstrtpxqtdepay.c:
- * gst/quicktime/gstrtpxqtdepay.h:
- * gst/quicktime/properties.c:
- * gst/quicktime/properties.h:
- * gst/quicktime/qtatomparser.h:
- * gst/quicktime/qtdemux.c:
- * gst/quicktime/qtdemux.h:
- * gst/quicktime/qtdemux.vcproj:
- * gst/quicktime/qtdemux_dump.c:
- * gst/quicktime/qtdemux_dump.h:
- * gst/quicktime/qtdemux_fourcc.h:
- * gst/quicktime/qtdemux_lang.c:
- * gst/quicktime/qtdemux_lang.h:
- * gst/quicktime/qtdemux_types.c:
- * gst/quicktime/qtdemux_types.h:
- * gst/quicktime/qtpalette.h:
- * gst/quicktime/quicktime.c:
* po/POTFILES.in:
quicktime: rename plugin to isomp4
https://bugzilla.gnome.org/show_bug.cgi?id=648004
* gst/dtmf/Makefile.am:
* gst/dtmf/gstdtmfcommon.h:
* gst/dtmf/gstdtmfsrc.c:
- * gst/dtmf/gstrtpdtmfcommon.h:
* gst/dtmf/gstrtpdtmfdepay.c:
* gst/dtmf/gstrtpdtmfdepay.h:
* gst/dtmf/gstrtpdtmfsrc.c:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/inspect/plugin-quicktime.xml:
- * gst/qtdemux/LEGAL:
- * gst/qtdemux/Makefile.am:
- * gst/qtdemux/gstrtpxqtdepay.c:
- * gst/qtdemux/gstrtpxqtdepay.h:
- * gst/qtdemux/qtatomparser.h:
- * gst/qtdemux/qtdemux.c:
- * gst/qtdemux/qtdemux.h:
- * gst/qtdemux/qtdemux.vcproj:
- * gst/qtdemux/qtdemux_dump.c:
- * gst/qtdemux/qtdemux_dump.h:
- * gst/qtdemux/qtdemux_fourcc.h:
- * gst/qtdemux/qtdemux_lang.c:
- * gst/qtdemux/qtdemux_lang.h:
- * gst/qtdemux/qtdemux_types.c:
- * gst/qtdemux/qtdemux_types.h:
- * gst/qtdemux/qtpalette.h:
- * gst/qtdemux/quicktime.c:
* gst/quicktime/LEGAL:
* gst/quicktime/Makefile.am:
* gst/quicktime/gstrtpxqtdepay.c:
2009-11-28 18:13:31 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* gst/aacparse/Makefile.am:
- * gst/aacparse/gstaacparse.c:
- * gst/aacparse/gstaacparse.h:
- * gst/aacparse/gstbaseparse.c:
- * gst/aacparse/gstbaseparse.h:
- * gst/aacparse/plugin.c:
* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
2009-05-15 17:11:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
- * tests/check/audiotestsrc.flac:
* tests/check/elements/id3demux.c:
* tests/check/elements/souphttpsrc.c:
* tests/check/pipelines/flacdec.c:
- * tests/check/test-cert.pem:
- * tests/check/test-key.pem:
* tests/files/Makefile.am:
* tests/files/audiotestsrc.flac:
* tests/files/test-cert.pem:
* gst/deinterlace/tvtime/weavebff.c:
* gst/deinterlace/tvtime/weavetff.c:
* gst/deinterlace/tvtime/x86-64_macros.inc:
- * gst/deinterlace2/Makefile.am:
- * gst/deinterlace2/gstdeinterlace2.c:
- * gst/deinterlace2/gstdeinterlace2.h:
- * gst/deinterlace2/tvtime/greedy.c:
- * gst/deinterlace2/tvtime/greedyh.asm:
- * gst/deinterlace2/tvtime/greedyh.c:
- * gst/deinterlace2/tvtime/greedyhmacros.h:
- * gst/deinterlace2/tvtime/linear.c:
- * gst/deinterlace2/tvtime/linearblend.c:
- * gst/deinterlace2/tvtime/mmx.h:
- * gst/deinterlace2/tvtime/plugins.h:
- * gst/deinterlace2/tvtime/scalerbob.c:
- * gst/deinterlace2/tvtime/sse.h:
- * gst/deinterlace2/tvtime/tomsmocomp.c:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoop0A.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopEdgeA8.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA2.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddA6.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopOddAH2.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVA.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/SearchLoopVAH.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
- * gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
- * gst/deinterlace2/tvtime/vfir.c:
- * gst/deinterlace2/tvtime/weave.c:
- * gst/deinterlace2/tvtime/weavebff.c:
- * gst/deinterlace2/tvtime/weavetff.c:
- * gst/deinterlace2/tvtime/x86-64_macros.inc:
Moved 'deinterlace2' from -bad to -good
And rename it to deinterlace.
* configure.ac:
* docs/plugins/Makefile.am:
- * gst/debug/Makefile.am:
- * gst/debug/breakmydata.c:
- * gst/debug/debug.vcproj:
- * gst/debug/efence.c:
- * gst/debug/efence.h:
- * gst/debug/efence.vcproj:
- * gst/debug/gstdebug.c:
- * gst/debug/gstnavigationtest.c:
- * gst/debug/gstnavigationtest.h:
- * gst/debug/gstnavseek.c:
- * gst/debug/gstnavseek.h:
- * gst/debug/gstpushfilesrc.c:
- * gst/debug/gstpushfilesrc.h:
- * gst/debug/gsttaginject.c:
- * gst/debug/gsttaginject.h:
- * gst/debug/navigationtest.vcproj:
- * gst/debug/negotiation.c:
- * gst/debug/progressreport.c:
- * gst/debug/progressreport.h:
- * gst/debug/rndbuffersize.c:
- * gst/debug/testplugin.c:
- * gst/debug/tests.c:
- * gst/debug/tests.h:
* gst/debugutils/Makefile.am:
* gst/debugutils/breakmydata.c:
* gst/debugutils/debug.vcproj:
2009-02-24 14:55:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/gconf/Makefile.am:
- * ext/gconf/gconf.c:
- * ext/gconf/gconf.h:
* ext/gconf/gstgconf.c:
* ext/gconf/gstgconf.h:
* ext/gconf/gstgconfelements.h:
* gst/audiofx/audioecho.c:
* gst/audiofx/audioecho.h:
* gst/audiofx/audiofx.c:
- * gst/audiofx/audioreverb.c:
- * gst/audiofx/audioreverb.h:
* tests/check/Makefile.am:
* tests/check/elements/audioecho.c:
- * tests/check/elements/audioreverb.c:
Rename audioreverb to audioecho. Fixes bug #568395.
The element can add an echo and a simple reverb effect to
an audio stream but for a real reverb filter it would need
projects / platform / upstream / gst-plugins-good.git / blobdiff