+=== release 1.9.1 ===
+
+2016-07-06 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.9.1
+
+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 Sebastian Dröge <slomo@coaxion.net>
+2016-03-24 12:27:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.8.0
+ * 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>
-# GStreamer 1.8 Release Notes
-
-**GStreamer 1.8.0 was released on 24 March 2016.**
-
-The GStreamer team is proud to announce a new major feature release in the
-stable 1.x API series of your favourite cross-platform multimedia framework!
-
-As always, this release is again packed with new features, bug fixes and other
-improvements.
-
-See [https://gstreamer.freedesktop.org/releases/1.8/][latest] for the latest
-version of this document.
-
-*Last updated: Thursday 24 March 2016, 10:00 UTC [(log)][gitlog]*
-
-[latest]: https://gstreamer.freedesktop.org/releases/1.8/
-[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.8/release-notes-1.8.md
-
-## Highlights
-
-- **Hardware-accelerated zero-copy video decoding on Android**
-
-- **New video capture source for Android using the android.hardware.Camera API**
-
-- **Windows Media reverse playback** support (ASF/WMV/WMA)
-
-- **New tracing system** provides support for more sophisticated debugging tools
-
-- **New high-level GstPlayer playback convenience API**
-
-- **Initial support for the new [Vulkan][vulkan] API**, see
- [Matthew Waters' blog post][vulkan-in-gstreamer] for more details
-
-- **Improved Opus audio codec support**: Support for more than two channels; MPEG-TS demuxer/muxer can now handle Opus;
- [sample-accurate][opus-sample-accurate] encoding/decoding/transmuxing with
- Ogg, Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container;
- [new codec utility functions for Opus header and caps handling][opus-codec-utils]
- in pbutils library. The Opus encoder/decoder elements were also moved to
- gst-plugins-base (from -bad), and the opus RTP depayloader/payloader to -good.
-
- [opus-sample-accurate]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiometa.html#GstAudioClippingMeta
- [opus-codec-utils]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstpbutilscodecutils.html
-
-- **GStreamer VAAPI module now released and maintained as part of the GStreamer project**
-
- [vulkan]: https://www.khronos.org/vulkan
- [vulkan-in-gstreamer]: http://ystreet00.blogspot.co.uk/2016/02/vulkan-in-gstreamer.html
-
-## Major new features and changes
-
-### Noteworthy new API, features and other changes
-
-- New GstVideoAffineTransformationMeta meta for adding a simple 4x4 affine
- transformation matrix to video buffers
-
-- [g\_autoptr()](https://developer.gnome.org/glib/stable/glib-Miscellaneous-Macros.html#g-autoptr)
- support for all types is exposed in GStreamer headers now, in combination
- with a sufficiently-new GLib version (i.e. 2.44 or later). This is primarily
- for the benefit of application developers who would like to make use of
- this, the GStreamer codebase itself will not be using g_autoptr() for
- the time being due to portability issues.
-
-- GstContexts are now automatically propagated to elements added to a bin
- or pipeline, and elements now maintain a list of contexts set on them.
- The list of contexts set on an element can now be queried using the new functions
- [gst\_element\_get\_context()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-context)
- and [gst\_element\_get\_contexts()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-contexts). GstContexts are used to share context-specific configuration objects
- between elements and can also be used by applications to set context-specific
- configuration objects on elements, e.g. for OpenGL or Hardware-accelerated
- video decoding.
-
-- New [GST\_BUFFER\_DTS\_OR\_PTS()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
- convenience macro that returns the decode timestamp if one is set and
- otherwise returns the presentation timestamp
-
-- New GstPadEventFullFunc that returns a GstFlowReturn instead of a gboolean.
- This new API is mostly for internal use and was added to fix a race condition
- where occasionally internal flow error messages were posted on the bus when
- sticky events were propagated at just the wrong moment whilst the pipeline
- was shutting down. This happened primarily when the pipeline was shut down
- immediately after starting it up. GStreamer would not know that the reason
- the events could not be propagated was because the pipeline was shutting down
- and not some other problem, and now the flow error allows GStreamer to know
- the reason for the failure (and that there's no reason to post an error
- message). This is particularly useful for queue-like elements which may need
- to asynchronously propagate a previous flow return from downstream.
-
-- Pipeline dumps in form of "dot files" now also show pad properties that
- differ from their default value, the same as it does for elements. This is
- useful for elements with pad subclasses that provide additional properties,
- e.g. videomixer or compositor.
-
-- Pad probes are now guaranteed to be called in the order they were added
- (before they were called in reverse order, but no particular order was
- documented or guaranteed)
-
-- Plugins can now have dependencies on device nodes (not just regular files)
- and also have a prefix filter. This is useful for plugins that expose
- features (elements) based on available devices, such as the video4linux
- plugin does with video decoders on certain embedded systems.
-
-- gst\_segment\_to\_position() has been deprecated and been replaced by the
- better-named gst\_segment\_position\_from\_running\_time(). At the same time
- gst\_segment\_position\_from\_stream\_time() was added, as well as \_full()
- variants of both to deal with negative stream time.
-
-- GstController: the interpolation control source gained a new monotonic cubic
- interpolation mode that, unlike the existing cubic mode, will never overshoot
- the min/max y values set.
-
-- GstNetAddressMeta: can now be read from buffers in language bindings as well,
- via the new gst\_buffer\_get\_net\_address\_meta() function
-
-- ID3 tag PRIV frames are now extraced into a new GST\_TAG\_PRIVATE\_DATA tag
-
-- gst-launch-1.0 and gst\_parse\_launch() now warn in the most common case if
- a dynamic pad link could not be resolved, instead of just silently
- waiting to see if a suitable pad appears later, which is often perceived
- by users as hanging -- they are now notified when this happens and can check
- their pipeline.
-
-- GstRTSPConnection now also parses custom RTSP message headers and retains
- them for the application instead of just ignoring them
-
-- rtspsrc handling of authentication over tunneled connections (e.g. RTSP over HTTP)
- was fixed
-
-- gst\_video\_convert\_sample() now crops if there is a crop meta on the input buffer
-
-- The debugging system printf functions are now exposed for general use, which
- supports special printf format specifiers such as GST\_PTR\_FORMAT and
- GST\_SEGMENT\_FORMAT to print GStreamer-related objects. This is handy for
- systems that want to prepare some debug log information to be output at a
- later point in time. The GStreamer-OpenGL subsystem is making use of these
- new functions, which are [gst\_info\_vasprintf()][gst_info_vasprintf],
- [gst\_info\_strdup\_vprintf()][gst_info_strdup_vprintf] and
- [gst\_info\_strdup\_printf()][gst_info_strdup_printf].
-
-- videoparse: "strides", "offsets" and "framesize" properties have been added to
- allow parsing raw data with strides and padding that do not match GStreamer
- defaults.
-
-- GstPreset reads presets from the directories given in GST\_PRESET\_PATH now.
- Presets are read from there after presets in the system path, but before
- application and user paths.
-
-[gst_info_vasprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-vasprintf
-[gst_info_strdup_vprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-vprintf
-[gst_info_strdup_printf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-printf
-
-### New Elements
-
-- [netsim](): a new (resurrected) element to simulate network jitter and
- packet dropping / duplication.
-
-- New VP9 RTP payloader/depayloader elements: rtpvp9pay/rtpvp9depay
-
-- New [videoframe_audiolevel]() element, a video frame synchronized audio level element
-
-- New spandsp-based tone generator source
-
-- New NVIDIA NVENC-based H.264 encoder for GPU-accelerated video encoding on
- suitable NVIDIA hardware
-
-- [rtspclientsink](), a new RTSP RECORD sink element, was added to gst-rtsp-server
-
-- [alsamidisrc](), a new ALSA MIDI sequencer source element
-
-### Noteworthy element features and additions
-
-- *identity*: new ["drop-buffer-flags"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-identity.html#GstIdentity--drop-buffer-flags)
- property to drop buffers based on buffer flags. This can be used to drop all
- non-keyframe buffers, for example.
-
-- *multiqueue*: various fixes and improvements, in particular special handling
- for sparse streams such as substitle streams, to make sure we don't overread
- them any more. For sparse streams it can be normal that there's no buffer for
- a long period of time, so having no buffer queued is perfectly normal. Before
- we would often unnecessarily try to fill the subtitle stream queue, which
- could lead to much more data being queued in multiqueue than necessary.
-
-- *multiqueue*/*queue*: When dealing with time limits, these elements now use the
- new ["GST_BUFFER_DTS_OR_PTS"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
- and ["gst_segment_to_running_time_full()"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstSegment.html#gst-segment-to-running-time-full)
- API, resulting in more accurate levels, especially when dealing with non-raw
- streams (where reordering happens, and we want to use the increasing DTS as
- opposed to the non-continuously increasing PTS) and out-of-segment input/output.
- Previously all encoded buffers before the segment start, which can happen when
- doing ACCURATE seeks, were not taken into account in the queue level calculation.
-
-- *multiqueue*: New ["use-interleave"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-multiqueue.html#GstMultiQueue--use-interleave)
- property which allows the size of the queues to be optimized based on the input
- streams interleave. This should only be used with input streams which are properly
- timestamped. It will be used in the future decodebin3 element.
-
-- *queue2*: new ["avg-in-rate"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-queue2.html#GstQueue2--avg-in-rate)
- property that returns the average input rate in bytes per second
-
-- audiotestsrc now supports all audio formats and is no longer artificially
- limited with regard to the number of channels or sample rate
-
-- gst-libav (ffmpeg codec wrapper): map and enable JPEG2000 decoder
-
-- multisocketsink can, on request, send a custom GstNetworkMessage event
- upstream whenever data is received from a client on a socket. Similarly,
- socketsrc will, on request, pick up GstNetworkMessage events from downstream
- and send any data contained within them via the socket. This allows for
- simple bidirectional communication.
-
-- matroska muxer and demuxer now support the ProRes video format
-
-- Improved VP8/VP9 decoding performance on multi-core systems by enabling
- multi-threaded decoding in the libvpx-based decoders on such systems
-
-- appsink has a new ["wait-on-eos"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-appsink.html#GstAppSink--wait-on-eos)
- property, so in cases where it is uncertain if an appsink will have a consumer for
- its buffers when it receives an EOS this can be set to FALSE to ensure that the
- appsink will not hang.
-
-- rtph264pay and rtph265pay have a new "config-interval" mode -1 that will
- re-send the setup data (SPS/PPS/VPS) before every keyframe to ensure
- optimal coverage and the shortest possibly start-up time for a new client
-
-- mpegtsmux can now mux H.265/HEVC video as well
-
-- The MXF muxer was ported to 1.x and produces more standard conformant files now
- that can be handled by more other software; The MXF demuxer got improved
- support for seek tables (IndexTableSegments).
-
-### Plugin moves
-
-- The rtph265pay/depay RTP payloader/depayloader elements for H.265/HEVC video
- from the rtph265 plugin in -bad have been moved into the existing rtp plugin
- in gst-plugins-good.
-
-- The mpg123 plugin containing a libmpg123 based audio decoder element has
- been moved from -bad to -ugly.
-
-- The Opus encoder/decoder elements have been moved to gst-plugins-base and
- the RTP payloader to gst-plugins-good, both coming from gst-plugins-bad.
-
-### New tracing tools for developers
-
-A new tracing subsystem API has been added to GStreamer, which provides
-external tracers with the possibility to strategically hook into GStreamer
-internals and collect data that can be evaluated later. These tracers are a
-new type of plugin features, and GStreamer core ships with a few example
-tracers (latency, stats, rusage, log) to start with. Tracers can be loaded
-and configured at start-up via an environment variable (GST\_TRACER\_PLUGINS).
-
-Background: While GStreamer provides plenty of data on what's going on in a
-pipeline via its debug log, that data is not necessarily structured enough to
-be generally useful, and the overhead to enable logging output for all data
-required might be too high in many cases. The new tracing system allows tracers
-to just obtain the data needed at the right spot with as little overhead as
-possible, which will be particularly useful on embedded systems.
-
-Of course it has always been possible to do performance benchmarks and debug
-memory leaks, memory consumption and invalid memory access using standard
-operating system tools, but there are some things that are difficult to track
-with the standard tools, and the new tracing system helps with that. Examples
-are things such as latency handling, buffer flow, ownership transfer of
-events and buffers from element to element, caps negotiation, etc.
-
-For some background on the new tracing system, watch Stefan Sauer's
-GStreamer Conference talk ["A new tracing subsystem for GStreamer"][tracer-0]
-and for a more specific example how it can be useful have a look at
-Thiago Santos's lightning talk ["Analyzing caps negotiation using GstTracer"][tracer-1]
-and his ["GstTracer experiments"][tracer-2] blog post. There was also a Google
-Summer of Code project in 2015 that used tracing system for a graphical
-GStreamer debugging tool ["gst-debugger"][tracer-3].
-
-This is all still very much work in progress, but we hope this will provide the
-foundation for a whole suite of new debugging tools for GStreamer pipelines.
-
-[tracer-0]: https://gstconf.ubicast.tv/videos/a-new-tracing-subsystem-for-gstreamer/
-[tracer-1]: https://gstconf.ubicast.tv/videos/analyzing-caps-negotiation-using-gsttracer/
-[tracer-2]: http://blog.thiagoss.com/2015/07/23/gsttracer-experiments/
-[tracer-3]: https://git.gnome.org/browse/gst-debugger
-
-### GstPlayer: a new high-level API for cross-platform multimedia playback
-
-GStreamer has had reasonably high-level API for multimedia playback
-in the form of the playbin element for a long time. This allowed application
-developers to just configure a URI to play, and playbin would take care of
-everything else. This works well, but there is still way too much to do on
-the application-side to implement a fully-featured playback application, and
-too much general GStreamer pipeline API exposed, making it less accessible
-to application developers.
-
-Enter GstPlayer. GstPlayer's aim is to provide an even higher-level abstraction
-of a fully-featured playback API but specialised for its specific use case. It
-also provides easy integration with and examples for Gtk+, Qt, Android, OS/X,
-iOS and Windows. Watch Sebastian's [GstPlayer talk at the GStreamer Conference][gstplayer-talk]
-for more information, or check out the [GstPlayer API reference][gstplayer-api]
-and [GstPlayer examples][gstplayer-examples].
-
-[gstplayer-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/player.html
-[gstplayer-talk]: https://gstconf.ubicast.tv/videos/gstplayer-a-simple-cross-platform-api-for-all-your-media-playback-needs-part-1/
-[gstplayer-examples]: https://github.com/sdroege/gst-player/
-
-### Adaptive streaming: DASH, HLS and MSS improvements
-
-- dashdemux now supports loading external xml nodes pointed from its MPD.
-
-- Content protection nodes parsing support for PlayReady WRM in mssdemux.
-
-- Reverse playback was improved to respect seek start and stop positions.
-
-- Adaptive demuxers (hlsdemux, dashdemux, mssdemux) now support the SNAP_AFTER
- and SNAP_BEFORE seek flags which will jump to the nearest fragment boundary
- when executing a seek, which means playback resumes more quickly after a seek.
-
-### Audio library improvements
-
-- audio conversion, quantization and channel up/downmixing functionality
- has been moved from the audioconvert element into the audio library and
- is now available as public API in form of [GstAudioConverter][audio-0],
- [GstAudioQuantize][audio-1] and [GstAudioChannelMixer][audio-2].
- Audio resampling will follow in future releases.
-
-- [gst\_audio\_channel\_get\_fallback\_mask()][audio-3] can be used
- to retrieve a default channel mask for a given number of channels as last
- resort if the layout is unknown
-
-- A new [GstAudioClippingMeta][audio-4] meta was added for specifying clipping
- on encoded audio buffers
-
-- A new GstAudioVisualizer base class for audio visualisation elements;
- most of the existing visualisers have been ported over to the new base class.
- This new base class lives in the pbutils library rather than the audio library,
- since we'd have had to make libgstaudio depend on libgstvideo otherwise,
- which was deemed undesirable.
-
-[audio-0]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-GstAudioConverter.html
-[audio-1]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-GstAudioQuantize.html
-[audio-2]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiochannels.html#gst-audio-channel-mix-new
-[audio-3]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiochannels.html#gst-audio-channel-get-fallback-mask
-[audio-4]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiometa.html#GstAudioClippingMeta
-
-### GStreamer OpenGL support improvements
-
-#### Better OpenGL Shader support
-
-[GstGLShader][shader] has been revamped to allow more OpenGL shader types
-by utilizing a new GstGLSLStage object. Each stage holds an OpenGL pipeline
-stage such as a vertex, fragment or a geometry shader that are all compiled
-separately into a program that is executed.
-
-The glshader element has also received a revamp as a result of the changes in
-the library. It does not take file locations for the vertex and fragment
-shaders anymore. Instead it takes the strings directly leaving the file
-management to the application.
-
-A new [example][liveshader-example] was added utilizing the new shader
-infrastructure showcasing live shader edits.
-
-[shader]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstglshader.html
-[liveshader-example]: https://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/tests/examples/gtk/glliveshader.c
-
-#### OpenGL GLMemory rework
-
-[GstGLMemory] was extensively reworked to support the addition of multiple
-texture targets required for zero-copy integration with the Android
-MediaCodec elements. This work was also used to provide IOSurface based
-GLMemory on OS X for zero-copy with OS X's VideoToolbox decoder (vtdec) and
-AV Foundation video source (avfvideosrc). There are also patches in bugzilla
-for GstGLMemoryEGL specifically aimed at improving the decoding performance on
-the Raspberry Pi.
-
-[GstGLMemory]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstglmemory.html
-
-A texture-target field was added to video/x-raw(memory:GLMemory) caps to signal
-the texture target contained in the GLMemory. Its values can be 2D, rectangle
-or external-oes. glcolorconvert can convert between the different formats as
-required and different elements will accept or produce different targets. e.g.
-glimagesink can take and render external-oes textures directly as required for
-effecient zero-copy on android.
-
-A generic GL allocation framework was also implemented to support the generic
-allocation of OpenGL buffers and textures which is used extensively by
-GstGLBufferPool.
-
-#### OpenGL DMABuf import uploader
-
-There is now a DMABuf uploader available for automatic selection that will
-attempt to import the upstream provided DMABuf. The uploader will import into
-2D textures with the necesarry format. YUV to RGB conversion is still provided
-by glcolorconvert to avoid the laxer restrictions with external-oes textures.
-
-#### OpenGL queries
-
-Queries of various aspects of the OpenGL runtime such as timers, number of
-samples or the current timestamp are not possible. The GstGLQuery object uses a
-delayed debug system to delay the debug output to later to avoid expensive calls
-to the glGet\* family of functions directly after finishing a query. It is
-currently used to output the time taken to perform various operations of texture
-uploads and downloads in GstGLMemory.
-
-#### New OpenGL elements
-
-glcolorbalance has been created mirroring the videobalance elements.
-glcolorbalance provides the exact same interface as videobalance so can be used
-as a GPU accelerated replacement. glcolorbalance has been added to glsinkbin so
-usage with playsink/playbin will use it automatically instead of videobalance
-where possible.
-
-glvideoflip, which is the OpenGL equiavalant of videoflip, implements the exact
-same interface and functionality as videoflip.
-
-#### EGL implementation now selects OpenGL 3.x
-
-The EGL implementation can now select OpenGL 3.x contexts.
-
-#### OpenGL API removal
-
-The GstGLDownload library object was removed as it was not used by anything.
-Everything is performed by GstGLMemory or in the gldownloadelement.
-
-The GstGLUploadMeta library object was removed as it was not being used and we
-don't want to promote the use of GstVideoGLTextureUploadMeta.
-
-#### OpenGL: Other miscellaneous changes
-
-- The EGL implementation can now select OpenGL 3.x contexts. This brings
- OpenGL 3.x to e.g. wayland and other EGL systems.
-
-- glstereomix/glstereosplit are now built and are usable on OpenGL ES systems
-
-- The UYVY/YUY2 to RGBA and RGBA to UYVY/YUY2 shaders were fixed removing the
- sawtooth pattern and luma bleeding.
-
-- We now utilize the GL\_APPLE\_sync extension on iOS devices which improves
- performance of OpenGL applications, especially with multiple OpenGL
- contexts.
-
-- glcolorconvert now uses a bufferpool to avoid costly
- glGenTextures/glDeleteTextures for every frame.
-
-- glvideomixer now has full glBlendFunc and glBlendEquation support per input.
-
-- gltransformation now support navigation events so your weird transformations
- also work with DVD menus.
-
-- qmlglsink can now run on iOS, OS X and Android in addition to the already
- supported Linux platform.
-
-- glimagesink now posts unhandled keyboard and mouse events (on backends that
- support user input, current only X11) on the bus for the application.
-
-### Initial GStreamer Vulkan support
-
-Some new elements, vulkansink and vulkanupload have been implemented utilizing
-the new Vulkan API. The implementation is currently limited to X11 platforms
-(via xcb) and does not perform any scaling of the stream's contents to the size
-of the available output.
-
-A lot of infrasctructure work has been undertaken to support using Vulkan in
-GStreamer in the future. A number of GstMemory subclasses have been created for
-integrating Vulkan's GPU memory handling along with VkBuffer's and VkImage's
-that can be passed between elements. Some GStreamer refcounted wrappers for
-global objects such as VkInstance, VkDevice, VkQueue, etc have also been
-implemented along with GstContext integration for sharing these objects with the
-application.
-
-### GStreamer VAAPI support for hardware-accelerated video decoding and encoding on Intel (and other) platforms
-
-#### GStreamer VAAPI is now part of upstream GStreamer
-
-The GStreamer-VAAPI module which provides support for hardware-accelerated
-video decoding, encoding and post-processing on Intel graphics hardware
-on Linux has moved from its previous home at the [Intel Open Source Technology Center][iostc]
-to the upstream GStreamer repositories, where it will in future be maintained
-as part of the upstream GStreamer project and released in lockstep with the
-other GStreamer modules. The current maintainers will continue to spearhead
-the development at the new location:
-
-[http://cgit.freedesktop.org/gstreamer/gstreamer-vaapi/][gst-vaapi-git]
-
-[gst-vaapi-git]: http://cgit.freedesktop.org/gstreamer/gstreamer-vaapi/
-
-GStreamer-VAAPI relies heavily on certain GStreamer infrastructure API that
-is still in flux such as the OpenGL integration API or the codec parser
-libraries, and one of the goals of the move was to be able to leverage
-new developments early and provide tighter integration with the latest
-developments of those APIs and other graphics-related APIs provided by
-GStreamer, which should hopefully improve performance even further and in
-some cases might also provide better stability.
-
-Thanks to everyone involved in making this move happen!
-
-#### GStreamer VAAPI: Bug tracking
-
-Bugs had already been tracked on [GNOME bugzilla](bgo) but will be moved
-from the gstreamer-vaapi product into a new gstreamer-vaapi component of
-the GStreamer product in bugzilla. Please file new bugs against the new
-component in the GStreamer product from now on.
-
-#### GStreamer VAAPI: Pending patches
-
-The code base has been re-indented to the GStreamer code style, which
-affected some files more than others. This means that some of the patches
-in bugzilla might not apply any longer, so if you have any unmerged patches
-sitting in bugzilla please consider checking if they still apply cleany and
-refresh them if not. Sorry for any inconvenience this may cause.
-
-#### GStreamer VAAPI: New versioning scheme and supported GStreamer versions
-
-The version numbering has been changed to match the GStreamer version
-numbering to avoid confusion: there is a new gstreamer-vaapi 1.6.0 release
-and a 1.6 branch that is roughly equivalent to the previous 0.7.0 version.
-Future releases 1.7.x and 1.8.x will be made alongside GStreamer releases.
-
-While it was possible and supported by previous releases to build against
-a whole range of different GStreamer versions (such as 1.2, 1.4, 1.6 or 1.7/1.8),
-in the future there will only be one target branch, so that git master will
-track GStreamer git master, 1.8.x will target GStreamer 1.8, and
-1.6.x will target the 1.6 series.
-
-[iostc]: http://01.org
-[bgo]: http://bugzilla.gnome.og
-
-#### GStreamer VAAPI: Miscellaneous changes
-
-All GStreamer-VAAPI functionality is now provided solely by its GStreamer
-elements. There is no more public library exposing GstVaapi API, this API
-was only ever meant for private use by the elements. Parts of it may be
-resurrected again in future if needed, but for now it has all been made
-private.
-
-GStreamer-VAAPI now unconditionally uses the codecparser library in
-gst-plugins-bad instead of shipping its own internal copy. Similarly,
-it no longer ships its own codec parsers but relies on the upstream
-codec parser elements.
-
-The GStreamer-VAAPI encoder elements have been renamed from vaapiencode_foo
-to vaapifooenc, so encoders are now called vaapih264enc, vaapih265enc,
-vaapimpeg2enc, vaapijpegenc, and vaapivp8enc. With this change we now follow
-the standard names in GStreamer, and the plugin documentation is generated
-correctly.
-
-In the case of the decoders, only the jpeg decoder has been split from the
-general decoding element vaapidecode: vaapijpegdec. This is the first step to
-split per codec each decoding element. The vaapijpegdec has also been given
-marginal rank for the time being.
-
-#### GStreamer VAAPI: New features in 1.8: 10-bit H.265/HEVC decoding support
-
-Support for decoding 10-bit H.265/HEVC has been added. For the time being
-this only works in combination with vaapisink though, until support for the
-P010 video format used internally is added to GStreamer and to the
-vaGetImage()/vaPutimage() API in the vaapi-intel-driver.
-
-Several fixes for memory leaks, build errors, and in the internal
-video parsing.
-
-Finally, vaapisink now posts the unhandled keyboard and mouse events to the
-application.
-
-### GStreamer Video 4 Linux Support
-
-Colorimetry support has been enhanced even more. It will now properly select
-default values when not specified by the driver. The range of color formats
-supported by GStreamer has been greatly improved. Notably, support for
-multi-planar I420 has been added along with all the new and non-ambiguous RGB
-formats that got added in recent kernels.
-
-The device provider now exposes a variety of properties as found in the udev
-database.
-
-The video decoder is now able to negotiate the downstream format.
-
-Elements that are dynamically created from /dev/video\* now track changes on
-these devices to ensure the registry stay up to date.
-
-All this and various bug fixes that improve both stability and correctness.
-
-### GStreamer Editing Services
-
-Added APIs to handle asset proxying support. Proxy creation is not the
-responsibility of GES itself, but GES provides all the needed features
-for it to be cleanly handled at a higher level.
-
-Added support for changing playback rate. This means that now, whenever a
-user adds a 'pitch' element (as it is the only known element to change playback
-rate through properties), GES will handle everything internally. This change
-introduced a new media-duration-factor property in NleObject which will
-lead to tweaking of seek events so they have the proper playback range to be
-requested upstream.
-
-Construction of NLE objects has been reworked making copy/pasting fully
-functional and allowing users to set properties on effects right after
-creating them.
-
-Rework of the title source to add more flexibility in text positioning,
-and letting the user get feedback about rendered text positioning.
-
-Report nlecomposition structural issues (coming from user programing mistakes)
-into ERROR messages on the bus.
-
-Add GI/pythyon testsuite in GES itself, making sure the API is working as expected
-in python, and allowing writing tests faster.
-
-### GstValidate
-
-Added support to run tests inside gdb.
-
-Added a 'smart' reporting mode where we give as much information as possible about
-critical errors.
-
-Uses GstTracer now instead of a LD\_PRELOAD library.
-
-## Miscellaneous
-
-- encodebin now works with "encoder-muxers" such as wavenc
-
-- gst-play-1.0 acquired a new keyboard shortcut: '0' seeks back to the start
-
-- gst-play-1.0 supports two new command line switches: -v for verbose output
- and --flags to configure the playbin flags to use.
-
-## Build and Dependencies
-
-- The GLib dependency requirement was bumped to 2.40
-
-- The -Bsymbolic configure check now works with clang as well
-
-- ffmpeg is now required as libav provider, incompatible changes were
- introduced that make it no longer viable to support both FFmpeg and Libav
- as libav providers. Most major distros have switched to FFmpeg or are in
- the process of switching to it anyway, so we don't expect this to be a
- problem, and there is still an internal copy of ffmpeg that can be used
- as fallback if needed.
-
-- The internal ffmpeg snapshot is now FFMpeg 3.0, but it should be possible
- to build against 2.8 as well for the time being.
-
-## Platform-specific improvements
-
-### Android
-
-- Zero-copy video decoding on Android using the hardware-accelerated decoders
- has been implemented, and is fully integrated with the GStreamer OpenGL stack
-
-- ahcsrc, a new camera source element, has been merged and can be used to
- capture video on android devices. It uses the android.hardware.Camera Java
- API to capture from the system's cameras.
-
-- The OpenGL-based QML video sink can now also be used on Android
-
-- New tinyalsasink element, which is mainly useful for Android but can also
- be used on other platforms.
-
-### OS/X and iOS
-
-- The system clock now uses mach\_absolute\_time() on OSX/iOS, which is
- the preferred high-resolution monotonic clock to be used on Apple platforms
-
-- The OpenGL-based QML video sink can now also be used on OS X and iOS (with
- some Qt build system massaging)
-
-- New IOSurface based memory implementation in avfvideosrc and vtdec on OS X
- for zerocopy with OpenGL. The previously used OpenGL extension
- GL_APPLE_ycbcr_422 is not compatible with GL 3.x core contexts.
-
-- New GstAppleCoreVideoMemory wrapping CVPixelBuffer's
-
-- avfvideosrc now supports renegotiation.
-
-### Windows
-
-- Various bugs with UDP and multicast were fixed on Windows, mostly related to
- gst-rtsp-server.
-
-- A few bugs in directsoundsrc and directsoundsink were fixed that could cause
- the element to lock up. Also the "mute" property on the sink was fixed, and
- a new "device" property for device selection was added to the source.
-
-## Known Issues
-
-- Building GStreamer applications with the Android NDK r11 is currently not
- supported due to incompatible changes in the NDK. This is expected to be
- fixed for 1.8.1.
- [Bugzilla #763999](https://bugzilla.gnome.org/show_bug.cgi?id=763999)
-
-- vp8enc crashes on 32 bit Windows, but was working fine in 1.6. 64 bit
- Windows is unaffected.
- [Bugzilla #763663](https://bugzilla.gnome.org/show_bug.cgi?id=763663)
-
-## Contributors
-
-Adam Miartus, Alban Bedel, Aleix Conchillo Flaqué, Aleksander Wabik,
-Alessandro Decina, Alex Ashley, Alex Dizengof, Alex Henrie, Alistair Buxton,
-Andreas Cadhalpun, Andreas Frisch, André Draszik, Anthony G. Basile,
-Antoine Jacoutot, Anton Bondarenko, Antonio Ospite, Arjen Veenhuizen,
-Arnaud Vrac, Arun Raghavan, Athanasios Oikonomou, Aurélien Zanelli, Ben Iofel,
-Bob Holcomb, Branko Subasic, Carlos Rafael Giani, Chris Bass, Csaba Toth,
-Daniel Kamil Kozar, Danilo Cesar Lemes de Paula, Dave Craig, David Fernandez,
-David Schleef, David Svensson Fors, David Waring, David Wu, Duncan Palmer,
-Edward Hervey, Egor Zaharov, Etienne Peron, Eunhae Choi, Evan Callaway,
-Evan Nemerson, Fabian Orccon, Florent Thiéry, Florin Apostol, Frédéric Wang,
-George Kiagiadakis, George Yunaev, Göran Jönsson, Graham Leggett,
-Guillaume Desmottes, Guillaume Marquebielle, Haihua Hu, Havard Graff,
-Heinrich Fink, Holger Kaelberer, HoonHee Lee, Hugues Fruchet, Hyunil Park,
-Hyunjun Ko, Ilya Konstantinov, James Stevenson, Jan Alexander Steffens (heftig),
-Jan Schmidt, Jason Litzinger, Jens Georg, Jimmy Ohn, Joan Pau Beltran,
-Joe Gorse, John Chang, John Slade, Jose Antonio Santos Cadenas, Josep Torra,
-Julian Bouzas, Julien Isorce, Julien Moutte, Justin Kim, Kazunori Kobayashi,
-Koop Mast, Lim Siew Hoon, Linus Svensson, Lubosz Sarnecki, Luis de Bethencourt,
-Lukasz Forynski, Manasa Athreya, Marcel Holtmann, Marcin Kolny, Marcus Prebble,
-Mark Nauwelaerts, Maroš Ondrášek, Martin Kelly, Matej Knopp, Mathias Hasselmann,
-Mathieu Duponchelle, Matt Crane, Matthew Marsh, Matthew Waters, Matthieu Bouron,
-Mersad Jelacic, Michael Olbrich, Miguel París Díaz, Mikhail Fludkov,
-Mischa Spiegelmock, Nicola Murino, Nicolas Dufresne, Nicolas Huet,
-Nirbheek Chauhan, Ognyan Tonchev, Olivier Crête, Pablo Anton, Pankaj Darak,
-Paolo Pettinato, Patricia Muscalu, Paul Arzelier, Pavel Bludov, Perry Hung,
-Peter Korsgaard, Peter Seiderer, Petr Viktorin, Philippe Normand,
-Philippe Renon, Philipp Zabel, Philip Van Hoof, Philip Withnall, Piotr Drąg,
-plamot, Polochon\_street, Prashant Gotarne, Rajat Verma, Ramiro Polla,
-Ravi Kiran K N, Reynaldo H. Verdejo Pinochet, Robert Swain, Romain Picard,
-Roman Nowicki, Ross Burton, Ryan Hendrickson, Santiago Carot-Nemesio,
-Scott D Phillips, Sebastian Dröge, Sebastian Rasmussen, Sergey Borovkov,
-Seungha Yang, Sjors Gielen, Song Bing, Sreerenj Balachandran, Srimanta Panda,
-Stavros Vagionitis, Stefan Sauer, Steven Hoving, Stian Selnes, Suhwang Kim,
-Thiago Santos, Thibault Saunier, Thijs Vermeir, Thomas Bluemel, Thomas Roos,
-Thomas Vander Stichele, Tim-Philipp Müller, Tim Sheridan, Ting-Wei Lan,
-Tom Deseyn, Vanessa Chipirrás Navalón, Víctor Manuel Jáquez Leal,
-Vincent Dehors, Vincent Penquerc'h, Vineeth T M, Vivia Nikolaidou,
-Wang Xin-yu (王昕宇), William Manley, Wim Taymans, Wonchul Lee, Xavi Artigas,
-Xavier Claessens, Youness Alaoui,
-
-... and many others who have contributed bug reports, translations, sent
-suggestions or helped testing.
-
-## Bugs fixed in 1.8
-
-More than [~700 bugs][bugs-fixed-in-1.8] have been fixed during
-the development of 1.8.
-
-This list does not include issues that have been cherry-picked into the
-stable 1.6 branch and fixed there as well, all fixes that ended up in the
-1.6 branch are also included in 1.8.
-
-This list also does not include issues that have been fixed without a bug
-report in bugzilla, so the actual number of fixes is much higher.
-
-[bugs-fixed-in-1.8]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=107311&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.6.1&target_milestone=1.6.2&target_milestone=1.6.3&target_milestone=1.7.0&target_milestone=1.7.1&target_milestone=1.7.2&target_milestone=1.7.3&target_milestone=1.7.4&target_milestone=1.7.90&target_milestone=1.7.91&target_milestone=1.7.92&target_milestone=1.7.x&target_milestone=1.8.0
-
-## Stable 1.8 branch
-
-After the 1.8.0 release there will be several 1.8.x bug-fix releases which
-will contain bug fixes which have been deemed suitable for a stable branch,
-but no new features or intrusive changes will be added to a bug-fix release
-usually. The 1.8.x bug-fix releases will be made from the git 1.8 branch, which
-is a stable branch.
-
-### 1.8.0
-
-1.8.0 was released on 24 March 2016.
-
-### 1.8.1
-
-The first 1.8 bug-fix release (1.8.1) is planned for April 2016.
-
-## Schedule for 1.10
-
-Our next major feature release will be 1.10, and 1.9 will be the unstable
-development version leading up to the stable 1.10 release. The development
-of 1.9/1.10 will happen in the git master branch.
-
-The plan for the 1.10 development cycle is yet to be confirmed, but it is
-expected that feature freeze will be around late July or early August,
-followed by several 1.9 pre-releases and the new 1.10 stable release
-in September.
-
-1.10 will be backwards-compatible to the stable 1.8, 1.6, 1.4, 1.2 and 1.0
-release series.
-
-- - -
-
-*These release notes have been prepared by Tim-Philipp Müller with
-contributions from Sebastian Dröge, Nicolas Dufresne, Edward Hervey, Víctor
-Manuel Jáquez Leal, Arun Raghavan, Thiago Santos, Thibault Saunier, Jan
-Schmidt and Matthew Waters.*
-
-*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)*
+This is GStreamer 1.9.1
projects / platform / upstream / gst-plugins-good.git / commitdiff