+=== 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 12:27:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.8.0
+
+2016-03-24 12:02:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-03-16 20:18:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: Use GstIterator for iterating all pads instead of manually iterating them while holding the object lock all the time
+ Doing queries while holding the object lock is a bit dangerous, and in this
+ case causes deadlocks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763326
+
+2016-03-17 20:53:27 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Fix typo to not change the input caps but our filtered caps
+ Changing the input caps and not using them anymore afterwards is useless, and
+ it breaks negotiation in pipelines like:
+ gst-launch-1.0 videotestsrc ! "video/x-raw,framerate=25/1,interlace-mode=interleaved" !
+ deinterlace fields=all ! "video/x-raw,framerate=50/1,interlace-mode=progressive" !
+ fakesink
+
+=== release 1.7.91 ===
+
+2016-03-15 12:04:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.91
+
+2016-03-15 11:53:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-03-15 11:41:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/hu.po:
+ * po/sr.po:
+ po: Update translations
+
+2016-03-15 03:26:14 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/rtpsource.c:
+ rtpmanager: Some comment and documentation clarifications/fixes
+
+2016-03-13 10:33:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ Revert "flacparse: push tags in pre_push_frame"
+ This reverts commit 4065fcb80a49924b70f0c8fc159dec0ff47943a1.
+ flacparse should not push tags by itself, the base class is going to do that
+ while properly merging in upstream tags. It just didn't because of a bug in
+ the base class, which was hidden by this commit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763553
+
+2016-02-25 05:17:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtp/dboolhuff.c:
+ * gst/rtp/dboolhuff.h:
+ * gst/rtp/gstrtpsbcpay.c:
+ win32: Don't use __attribute__ on MSVC
+ Use MSVC-equivalents for alignment and packing compiler directives when building
+ on MSVC
+
+2016-02-25 05:16:42 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/matroska/ebml-read.c:
+ win32: Don't try to include xmath.h on newer Visual Studio
+
+2016-02-25 05:16:09 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ * gst/goom/gstgoom.c:
+ * gst/goom2k1/gstgoom.c:
+ * gst/monoscope/gstmonoscope.c:
+ gst Factor out endian-order RGB formats
+ MSVC seems to ignore preprocessor conditionals inside static pad
+ template macros.
+
+2016-03-08 17:37:17 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ dirctsoundsink: Setting volume should not unmute
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:57:24 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ dirctsoundsink: Fix volume reset on unmute
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2016-03-08 13:03:55 +0100 Alban Bedel <alban.bedel@avionic-design.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: fix capture with bayer formats other than bggr
+ gst_v4l2_object_get_caps_info() always return V4L2_PIX_FMT_SBGGR8
+ for all bayer formats. This is obviously broken if the device use
+ another ordering. Fix this by properly reading the format parameter.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763318
+
+2016-03-07 10:28:06 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: reset pending segment if we are already pushing one
+ When upstream is running in bytes in push-mode, qtdemux will
+ convert seeks from time to bytes and send it upstream. Upstream
+ element will perform a byte seek and send a byte segment to qtdemux
+ that will convert it to time and push it downstream.
+ There is, however, the pending_segment variable that stores a new
+ segment event to be pushed before the next data. When handling seeks
+ as mentioned above this variable was being ignored and, if it contained
+ some segment event, it would override the one resulting from the seek.
+ This would restore a previous segment and would cause the seek segment
+ to be discarded downstream.
+ This patch fixes this issue by unrefing any pending segment as the
+ seek from upstream should contain the latest one that should be
+ used, as requested by the application.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763165
+
+2016-03-07 10:27:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: run gst-indent
+ Otherwise commits will fail with our indent check hook
+
+2016-03-04 15:09:45 +0100 Josep Torra <n770galaxy@gmail.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: fix colorimetry for NV12
+ Replicate V4L2_MAP_QUANTIZATION_DEFAULT macro behavior.
+ At #v4l it was described that documentation might be wrong and that
+ we should trust this macro instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762529
+
+2016-03-04 14:07:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix multicast group joining with provided sockets on Windows
+ On Windows the socket will be bound to ANY instead of the multicast group,
+ as binding to a multicast group does not work. Which would mean that we
+ override src->addr to become ANY and won't automatically join a multicast
+ group anymore on Windows.
+ On Linux we would automatically join a multicast group, keep it consistent.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763093
+
+2016-03-02 13:13:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases"
+ This reverts commit a7fb7b53592d87f7983544debb74d364fc3257ad.
+ The mutex is taken by the caller, we should keep it locked when returning so
+ the caller can unlock it again.
+
+2016-03-01 15:01:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: push tags in pre_push_frame
+ Push a tag event before pre-roll if we have tags.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762660
+
+=== release 1.7.90 ===
+
+2016-03-01 18:15:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.90
+
+2016-03-01 17:03:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/ca.po:
+ * po/da.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/or.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/tr.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2016-03-01 16:53:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/bg.po:
+ * po/cs.po:
+ * po/de.po:
+ * po/fr.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2016-03-01 14:14:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: don't forget to unlock mutex in error code path in two cases
+
+2016-02-29 10:10:24 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: remove impossible condition
+ It is impossible for a guint to have a negative value, no need to check for
+ this. Introduced in commit 6861d11c49ea0f30d2432cf4ebf6108bc89897f1
+ CID 1354509
+
+2016-02-28 10:12:36 +0100 Petr Viktorin <encukou@gmail.com>
+
+ * gst/alpha/gstalpha.c:
+ alpha: Fix sample pipeline
+ Use the zorder pad property to make sure the semitransparent
+ video is on top of the background.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762809
+
+2016-02-28 13:42:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/replaygain/gstrgvolume.c:
+ * tests/check/elements/rgvolume.c:
+ rgvolume: make tag list writable before modifying it
+ Making the event itself writable is not enough, it won't make
+ the actual taglist in the event writable as well. Instead, just
+ make a copy of the taglist and then create a new tag event from
+ that if required, replacing the old one. Before we would
+ inadvertently modify taglists upstream elements might still
+ be holding on to. Add unit test for this as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762793
+
+2016-02-28 13:01:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Properly error out if binding the UDP sockets fails
+ udpsrc is not returning us a socket in that case.
+
+2016-02-27 20:33:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ goom: Use goom_set_resolution() instead of recreating the goom instance when the resolution changes
+ https://bugzilla.gnome.org/show_bug.cgi?id=762765
+
+2016-02-27 20:32:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ Revert "goom: Initialize the goom struct only once we know width/height and recreate it if those change"
+ This reverts commit cc6e102643c1bae928316dca9f34db028fb9a67e.
+
+2016-02-27 20:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ goom: Initialize the goom struct only once we know width/height and recreate it if those change
+ Fixes crash when the width and/or height is changing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762765
+
+2016-02-26 12:41:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b64f03f to 6f2d209
+
+2016-02-25 22:54:18 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ docs: add rtpopusdepay and rtpopuspay to documentation
+
+2016-02-17 15:15:11 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ rtp: opus: move Opus RTP payloader/depayloader from -bad to -good
+ https://bugzilla.gnome.org/show_bug.cgi?id=756282
+
+2016-02-17 15:10:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ Merge branch 'plugin-move-rtp-opus'
+ Move Opus RTP depayloader/payloader from -bad to -good.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756282
+
+2016-02-25 11:33:13 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: cenc aux info parsing from mdat support in PULL mode
+ This is already supported for PUSH mode but was failing in PULL mode.
+ The aux info is sometimes stored in the mdat before the first sample,
+ so the loop task needs to pull data stored at that location and
+ perform the aux info cenc parsing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761700
+ https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2016-02-24 11:28:09 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: prevent buffer flow if any stream failed to be exposed
+ In some cases the stream configuration can fail, for instance if the
+ stream is protected and no decryptor was found. For those situations
+ the demuxer shouldn't emit any data on the corresponding source pad of
+ the stream and bail out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2016-02-24 09:12:03 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: don't push encrypted buffer without cenc metadata
+ When the cenc metadata is stored outside of the moof box and the
+ stream is exposed it is possible that the cenc metadata hasn't been
+ processed yet while the first buffer is being pushed. When this
+ happens the buffer can't possibly be decrypted downstream so don't
+ push it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762516
+
+2015-10-21 16:21:45 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: read saio aux_info_type as a FOURCC
+ https://bugzilla.gnome.org/show_bug.cgi?id=756897
+
+2016-02-23 18:27:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdec.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/smpte/gstsmpte.c:
+ gst: Handle gst_pad_get_current_caps() returning NULL gracefully
+
+2016-02-23 18:12:54 +0200 Dave Craig <dcraig@brightsign.biz>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
+ Remove calls to gst_pad_has_current_caps() which then go on to call
+ gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
+ use gst_pad_get_current_caps() and check for NULL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2015-12-16 12:40:39 +0000 Dave Craig <dcraig@brightsign.biz>
+
+ * ext/flac/gstflacenc.c:
+ * gst/flv/gstflvmux.c:
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/shapewipe/gstshapewipe.c:
+ * gst/videocrop/gstaspectratiocrop.c:
+ gst: Don't assume that get_current_caps() returns non-NULL caps after has_current_caps()
+ Remove calls to gst_pad_has_current_caps() which then go on to call
+ gst_pad_get_current_caps() as the caps can go to NULL in between. Instead just
+ use gst_pad_get_current_caps() and check for NULL.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2015-12-16 10:54:17 +0000 Dave Craig <dcraig@brightsign.biz>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Handle gst_pad_get_current_caps() returning NULL gracefully
+ This can happen when the pipeline is currently shutting down.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759539
+
+2016-02-23 15:57:18 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Don't handle seek until ready
+ https://bugzilla.gnome.org/show_bug.cgi?id=762542
+
+2016-02-23 15:55:13 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Unref seek event
+ https://bugzilla.gnome.org/show_bug.cgi?id=762542
+
+2016-02-22 11:01:40 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: close file on write error with next-file mode is set to buffer
+ If we have an error during fwrite call, file stays open and thus next
+ incoming buffer will trigger an assert when trying to opening a new
+ file.
+ This happens if we do not restart element, file is closed at stop, and
+ if application handles the returned GST_FLOW_ERROR to keep bin alive.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762434
+
+2016-02-19 23:44:42 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: don't output empty tags/tag elements
+ Such files will not play on Android, because of bug in libwebm matroska parsing, which is still present in 6.0.1
+ https://bugzilla.gnome.org/show_bug.cgi?id=762349
+
+2016-02-04 15:59:04 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: make up an OpusHead block if possible when missing
+ https://bugzilla.gnome.org/show_bug.cgi?id=761489
+
+2016-02-04 10:43:15 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: make up an OpusHead block if possible when missing
+ This block is needed in the Matroska file, but data coming from
+ RTP may not have one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761489
+
+2016-02-22 13:53:21 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: make stream-id more readable and order-friendly
+ ... as streams are so ordered by id by e.g. decodebin
+ (and as typically already honoured by other demuxers).
+
+2016-02-22 13:25:51 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: remove confusing duplicate track uid field
+
+2016-02-22 14:03:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtpvp9pay.c:
+ rtpvp9pay: add missing break
+ VP9_PAY_PICTURE_ID_7BITS and VP9_PAY_PICTURE_ID_15BITS are mutually
+ exclusive options of the picture-id-mode. We can break after the
+ first case.
+ 1 or 2 bytes need to be added to the header length depending on the
+ PictureID size.
+ https://tools.ietf.org/html/draft-uberti-payload-vp9-00#section-4.2
+ CID 1353479
+
+2016-02-22 09:09:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix buffer memory leak
+ buffer being mapped is not being unmapped in some cases
+ https://bugzilla.gnome.org/show_bug.cgi?id=762420
+
+2015-11-04 10:19:03 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpmanager: Don't warn for duplicate/reordered packets
+ This is a normal scenario and should not be a warning.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762208
+
+2016-02-21 09:47:43 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/alpha/alpha.vcproj:
+ * gst/auparse/auparse.vcproj:
+ * gst/avi/avi.vcproj:
+ * gst/cutter/cutter.vcproj:
+ * gst/debugutils/debug.vcproj:
+ * gst/debugutils/navigationtest.vcproj:
+ * gst/effectv/effectv.vcproj:
+ * gst/flx/flxdec.vcproj:
+ * gst/goom/goom.vcproj:
+ * gst/goom2k1/goom.vcproj:
+ * gst/interleave/interleave.vcproj:
+ * gst/isomp4/qtdemux.vcproj:
+ * gst/law/alaw.vcproj:
+ * gst/law/mulaw.vcproj:
+ * gst/matroska/matroska.vcproj:
+ * gst/multipart/multipart.vcproj:
+ * gst/rtp/rtp.vcproj:
+ * gst/smpte/smpte.vcproj:
+ * gst/spectrum/spectrum.vcproj:
+ * gst/udp/udp.vcproj:
+ * gst/videobox/videobox.vcproj:
+ * gst/videocrop/videocrop.vcproj:
+ * gst/videofilter/gamma.vcproj:
+ * gst/videofilter/videobalance.vcproj:
+ * gst/videofilter/videofilter.vcproj:
+ * gst/videofilter/videoflip.vcproj:
+ * gst/videomixer/videomixer.vcproj:
+ * gst/wavenc/wavenc.vcproj:
+ * gst/wavparse/wavparse.vcproj:
+ * gst/y4m/y4menc.vcproj:
+ * win32/MANIFEST:
+ * win32/vs6/autogen.dsp:
+ * win32/vs6/gst_plugins_good.dsw:
+ * win32/vs6/libgstalaw.dsp:
+ * win32/vs6/libgstalpha.dsp:
+ * win32/vs6/libgstalphacolor.dsp:
+ * win32/vs6/libgstapetag.dsp:
+ * win32/vs6/libgstaudiofx.dsp:
+ * win32/vs6/libgstauparse.dsp:
+ * win32/vs6/libgstautodetect.dsp:
+ * win32/vs6/libgstavi.dsp:
+ * win32/vs6/libgstcutter.dsp:
+ * win32/vs6/libgstdirectsound.dsp:
+ * win32/vs6/libgsteffectv.dsp:
+ * win32/vs6/libgstflx.dsp:
+ * win32/vs6/libgstgoom.dsp:
+ * win32/vs6/libgsticydemux.dsp:
+ * win32/vs6/libgstid3demux.dsp:
+ * win32/vs6/libgstinterleave.dsp:
+ * win32/vs6/libgstjpeg.dsp:
+ * win32/vs6/libgstlevel.dsp:
+ * win32/vs6/libgstmatroska.dsp:
+ * win32/vs6/libgstmedian.dsp:
+ * win32/vs6/libgstmonoscope.dsp:
+ * win32/vs6/libgstmulaw.dsp:
+ * win32/vs6/libgstmultipart.dsp:
+ * win32/vs6/libgstpng.dsp:
+ * win32/vs6/libgstqtdemux.dsp:
+ * win32/vs6/libgstrtp.dsp:
+ * win32/vs6/libgstrtsp.dsp:
+ * win32/vs6/libgstsmpte.dsp:
+ * win32/vs6/libgstspeex.dsp:
+ * win32/vs6/libgstudp.dsp:
+ * win32/vs6/libgstvideobalance.dsp:
+ * win32/vs6/libgstvideobox.dsp:
+ * win32/vs6/libgstvideocrop.dsp:
+ * win32/vs6/libgstvideoflip.dsp:
+ * win32/vs6/libgstvideomixer.dsp:
+ * win32/vs6/libgstwaveform.dsp:
+ * win32/vs6/libgstwavenc.dsp:
+ * win32/vs6/libgstwavparse.dsp:
+ * win32/vs7/libgstdirectsound.vcproj:
+ * win32/vs8/gst-plugins-good.sln:
+ * win32/vs8/libgst1394.vcproj:
+ * win32/vs8/libgstaasink.vcproj:
+ * win32/vs8/libgstalaw.vcproj:
+ * win32/vs8/libgstalpha.vcproj:
+ * win32/vs8/libgstalphacolor.vcproj:
+ * win32/vs8/libgstannodex.vcproj:
+ * win32/vs8/libgstapetag.vcproj:
+ * win32/vs8/libgstaudiofx.vcproj:
+ * win32/vs8/libgstauparse.vcproj:
+ * win32/vs8/libgstautodetect.vcproj:
+ * win32/vs8/libgstavi.vcproj:
+ * win32/vs8/libgstcacasink.vcproj:
+ * win32/vs8/libgstcdio.vcproj:
+ * win32/vs8/libgstcutter.vcproj:
+ * win32/vs8/libgstdirectsound.vcproj:
+ * win32/vs8/libgstdv.vcproj:
+ * win32/vs8/libgsteffectv.vcproj:
+ * win32/vs8/libgstflac.vcproj:
+ * win32/vs8/libgstflxdec.vcproj:
+ * win32/vs8/libgstgoom.vcproj:
+ * win32/vs8/libgsticydemux.vcproj:
+ * win32/vs8/libgstid3demux.vcproj:
+ * win32/vs8/libgstjpeg.vcproj:
+ * win32/vs8/libgstladspa.vcproj:
+ * win32/vs8/libgstlevel.vcproj:
+ * win32/vs8/libgstmatroska.vcproj:
+ * win32/vs8/libgstmng.vcproj:
+ * win32/vs8/libgstmonoscope.vcproj:
+ * win32/vs8/libgstmulaw.vcproj:
+ * win32/vs8/libgstmultipart.vcproj:
+ * win32/vs8/libgstpng.vcproj:
+ * win32/vs8/libgstrtp.vcproj:
+ * win32/vs8/libgstrtsp.vcproj:
+ * win32/vs8/libgstshout2.vcproj:
+ * win32/vs8/libgstsmpte.vcproj:
+ * win32/vs8/libgstspeex.vcproj:
+ * win32/vs8/libgsttaglib.vcproj:
+ * win32/vs8/libgstudp.vcproj:
+ * win32/vs8/libgstvideobalance.vcproj:
+ * win32/vs8/libgstvideobox.vcproj:
+ * win32/vs8/libgstvideoflip.vcproj:
+ * win32/vs8/libgstvideomixer.vcproj:
+ * win32/vs8/libgstwavenc.vcproj:
+ * win32/vs8/libgstwavparse.vcproj:
+ win32: remove outdated build cruft
+ This hasn't been touched for generations, doesn't work,
+ and is just causing confusion. We also don't want to
+ maintain these files manually.
+
+2016-02-20 11:51:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: don't use undeclared core debug category symbols
+
+2016-02-06 14:39:05 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: workaround for files with wrong color_table_id value
+ Instead of erroring out, just use the default color table.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761637
+
+2016-02-19 15:02:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/rtp/gstrtpvp9depay.c:
+ flvmux, rtpvp9depay: fix indentation
+
+2016-02-19 15:03:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2src: fix indentation
+
+2015-12-04 00:46:34 +1100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: plug leak(s) in error-scenario
+ https://bugzilla.gnome.org/show_bug.cgi?id=762210
+
+2015-12-04 00:46:12 +1100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix eos event leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=762209
+
+2016-02-19 14:41:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/flvdemux.c:
+ * tests/check/elements/flvmux.c:
+ * tests/check/elements/rtph263.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: fix indentation
+
+2016-02-18 16:09:29 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: port testharness to GstHarness and cleanup/improve
+ Probably found a bug as well, in that there are some timestamps in
+ there that are looking very wrong. (marked with FIXME)
+ https://bugzilla.gnome.org/show_bug.cgi?id=762267
+
+2016-02-18 10:27:19 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: test cleanups/improvements
+ Use fail_unless and friends instead of g_assert
+ Factor seq-num checking out to separate function
+ Check more return-values from push and crank and others
+ https://bugzilla.gnome.org/show_bug.cgi?id=762254
+
+2015-12-03 11:07:05 +0100 Stian Selnes <stian@pexip.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: fix leaks in unit test
+ https://bugzilla.gnome.org/show_bug.cgi?id=762214
+
+2016-02-19 12:38:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.2 ===
+
+2016-02-19 11:49:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * 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:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.2
+
+2016-02-19 10:31:48 +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:
+ po: Update translations
+
+2016-02-18 18:33:13 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: plug leaks in cenc aux info parsing
+
+2016-02-18 13:43:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: fix spurious souphttpsrc test timouts
+ Set GSETTINGS_BACKEND=memory, apparently there's something
+ about fork() and the dconf backend (or whatever else that
+ drags in or activates) that messes up locking and causes
+ timeouts due to deadlocks in g_mutex_lock(), since
+ everything works fine with CK_FORK=no as well.
+
+2016-02-18 11:10:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Unmap wavpack header buffer after creating it
+ Otherwise it will be mapped writable all the time and we can't read from it
+ anywhere.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762239
+
+2015-12-08 18:49:40 +0100 Stian Selnes <stian@pexip.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Add test for big seqnum gap handling
+ Make sure that the packets queued when detecting a big gap are pushed
+ after reset (5 consective seqnums) and not dropped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762211
+
+2016-02-17 15:03:13 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtputils.h:
+ rtp: sprinkle some G_GNUC_INTERNAL for internal utils functions
+
+2016-02-09 13:17:00 +0000 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only transform protected caps once
+ Commit 7873bede3134b15e5066e8d14e54d1f5054d2063
+ (https://bugzilla.gnome.org/show_bug.cgi?id=760774) changed the
+ behaviour of qtdemux to call gst_qtdemux_configure_stream() for
+ every new moof.
+ When playing a protected stream, gst_qtdemux_configure_stream()
+ calls gst_qtdemux_configure_protected_caps(). The
+ gst_qtdemux_configure_protected_caps() function takes the original
+ media format, puts this in a field called "original-media-type"
+ and then changes the caps to "application/x-cenc".
+ The gst_qtdemux_configure_protected_caps() did not handle the case
+ of being called multiple times, causing it to incorrectly set the
+ caps. The second call was causing the caps to be set to:
+ application/x-cenc, original-media-type"application/x-cenc"
+ This commit makes gst_qtdemux_configure_protected_caps() check that
+ the caps have already been transformed, so that it only gets
+ changed once.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761769
+
+2015-11-03 14:50:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ opus: Add proper support for multichannel audio
+ https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-06-30 13:51:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ opus: Copy metadata in the (de)payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without tags or
+ with only the audio tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-05-04 11:23:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ opusdepay: Set multistream=FALSE on the Opus caps
+ The RTP Opus mapping only allows mono/stereo, and not multistream Opus
+ streams.
+
+2015-03-24 13:57:54 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: Forward stereo preferences from caps upstream
+ https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-03-24 13:56:21 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: Set the number of channels to 2 as per RFC draft
+ https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-03-23 12:24:55 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ opus: Handle sprop-stereo and sprop-maxcapturerate RTP caps fields
+ https://bugzilla.gnome.org/show_bug.cgi?id=746617
+
+2015-02-19 14:30:10 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: default encoding name to OPUS
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-19 14:05:06 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: make caps writable before truncating them
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-05 10:27:51 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: negotiate the encoding name
+ Chrome uses a different encoding name that gstreamer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2014-11-01 10:10:27 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopus: Use OPUS encoding name
+ Both Firefox and Chrome uses OPUS as the encoding in their SDP.
+ Adding this now defacto standard name remove the need for special
+ case in SDP parsing code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2013-01-31 12:30:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opuspay: fix timestamps
+ Copy timestamps to payloaded buffer.
+ Avoid input buffer memory leak.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929
+
+2012-11-03 20:38:00 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ Fix FSF address
+ https://bugzilla.gnome.org/show_bug.cgi?id=687520
+
+2012-10-22 12:08:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opuspay: remove pointless caps serialization
+ Remove the caps serialization in the rtp caps. the spec nor the receiver
+ does anything with it.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686547
+
+2012-10-17 17:34:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ Use gst_element_class_set_static_metadata()
+ where possible. Avoids some string copies. Also re-indent
+ some stuff. Also some indent fixes here and there.
+
+2012-09-20 18:41:24 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: Allocate the rtp buffer correctly
+ Use the right functions to allocate the rtp buffer
+
+2012-09-14 17:08:49 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopuspay.c:
+ replace gst_element_class_set_details_simple with gst_element_class_set_metadata
+
+2012-03-07 17:14:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opus: port to updated 0.11
+
+2011-12-30 11:41:17 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ Merge remote-tracking branch 'origin/master' into 0.11-premerge
+ Conflicts:
+ docs/libs/Makefile.am
+ ext/kate/gstkatetiger.c
+ ext/opus/gstopusdec.c
+ ext/xvid/gstxvidenc.c
+ gst-libs/gst/basecamerabinsrc/Makefile.am
+ gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
+ gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
+ gst-libs/gst/video/gstbasevideocodec.c
+ gst-libs/gst/video/gstbasevideocodec.h
+ gst-libs/gst/video/gstbasevideodecoder.c
+ gst-libs/gst/video/gstbasevideoencoder.c
+ gst/asfmux/gstasfmux.c
+ gst/audiovisualizers/gstwavescope.c
+ gst/camerabin2/gstcamerabin2.c
+ gst/debugutils/gstcompare.c
+ gst/frei0r/gstfrei0rmixer.c
+ gst/mpegpsmux/mpegpsmux.c
+ gst/mpegtsmux/mpegtsmux.c
+ gst/mxf/mxfmux.c
+ gst/videomeasure/gstvideomeasure_ssim.c
+ gst/videoparsers/gsth264parse.c
+ gst/videoparsers/gstmpeg4videoparse.c
+
+2011-12-09 17:25:41 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpopuspay.c:
+ opusenc: add upstream negotiation for multistream ability
+ This will help elements that cannot deal with multistream,
+ such as the RTP payloader.
+ The caps now do not include a "streams" field anymore, but
+ a "multistream" boolean, since we have no real use for knowing
+ the exact amount of streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=665078
+
+2011-12-07 15:13:11 -0200 Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ Adding opus RTP payloader/depayloader element
+ Adding OPUS RTP module based on the current draft:
+ http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt
+ https://bugzilla.gnome.org/show_bug.cgi?id=664817
+
+2016-02-17 13:26:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtputils.c:
+ * gst/rtp/gstrtputils.h:
+ rtp: h264/h265: avoid duplication of read_golomb()
+ There is no need to have two identical implementations of the read_golomb
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-17 14:37:44 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Simple implementation of TRICKMODE_KEY_UNITS
+ When the trickmode key-units flag is set on the segment, simply skip
+ any sample on a video stream that isn't a keyframe
+ https://bugzilla.gnome.org/show_bug.cgi?id=762185
+
+2015-08-21 14:15:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: send GAP events for lagging audio and video streams too
+ Send GAP events for non-subtitle streams too if they lag too much
+ behind, but use a higher threshold than for subtitles.
+ This helps with fixing prerolling with a file where one of the
+ audio streams only has data starting from 19s onwards. It's not
+ a complete fix yet, it also requires changes elsewhere, such as
+ in baseparse, to make sure caps are propagated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=614460
+ https://bugzilla.gnome.org/show_bug.cgi?id=753899
+
+2015-12-23 19:54:13 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpvp9depay.c:
+ * gst/rtp/gstrtpvp9depay.h:
+ * gst/rtp/gstrtpvp9pay.c:
+ * gst/rtp/gstrtpvp9pay.h:
+ rtpvp9pay: rtpvp9depay: Initial implementation of draft 01
+ Quick and dirty implementation of an RTP payloader and depayloader
+ for VP9. In particalur it assumes no spatial or temporal layering,
+ non-flexible mode, and some other bits and pieces.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754773
+
+2016-02-16 09:02:30 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix string memory leak
+ codec_name is not being freed in all conditions leading to memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=762117
+
+2015-12-10 12:15:52 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: add "get-session" signal
+ This gets the GstRTPSession element, as compared to the RTPSession object
+ that is returned by get-internal-session.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759293
+
+2016-02-16 00:19:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ rtp: h265: hook up move RTP H.265 payloader/depayloader to build
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-16 00:14:27 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ rtp: h265: use common meta utility functions
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-05 18:18:31 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.h:
+ * gst/rtp/gstrtph265types.h:
+ rtp: h265: remove codecparser dependency from h265 payloader/depayloader
+ Looks like it just uses the NAL enums and nothing else from
+ the codecparsers, and that's the only reason it had to be
+ moved from -good to -bad when it was originally added. We
+ can probably keep those NAL enums up to date enough, so let's
+ remove the codecparser dependency so it can be moved back into
+ -good.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-16 00:24:58 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ Merge branch 'plugin-move-rtp-h265'
+ Move RTP H.265 payloader/depayloader from -bad to -good.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761606
+
+2016-02-05 15:34:51 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ gstrtph265depay: keep consistency with rtph264depay
+ Use gst_rtp_drop_meta() and the same function prototype for
+ gst_rtp_copy_meta() to keep consistency with the RTP elements in
+ gst-plugins-good
+
+2016-02-05 13:56:34 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: fix termination of access unit
+ Only consider the access unit complete when the next-occurring VCL NAL unit
+ has the first bit after its NAL unit header equal to 1.
+
+2016-01-15 16:10:02 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: fix unneeded sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then throw it
+ away immediately, just use the original input buffer directly.
+
+2016-01-15 15:56:59 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: add "send VPS/SPS/PPS with every key frame" mode
+ It's not enough to have timeout or event based VPS/SPS/PPS information
+ sent in RTP packets. There are some scenarios when key frames may appear
+ more frequently than once a second, in which case the minimum timeout
+ for "config-interval" of 1 second for sending VPS/SPS/PPS isn't enough.
+ It might also be desirable in general to make sure the VPS/SPS/PPS is
+ available with every keyframe (packet loss aside), so receivers can
+ actually pick up decoding immediately from the first keyframe if
+ VPS/SPS/PPS is not signaled out of band.
+ This commit adds the possibility to send VPS/SPS/PPS with every key frame.
+ This mode can be enabled by setting "config-interval" property to -1. In
+ this case the payloader will add VPS, SPS and PPS before every key (IDR)
+ frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2016-01-15 15:19:41 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtph265pay: change config-interval property type from uint to int
+ This way we can use -1 as special value, which is nicer than MAXUINT.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-08-15 16:22:20 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: make sure we call handle_nal for each NAL
+ Call handle_nal for each NAL in the STAP-A RTP packet. This makes sure
+ we correctly extract the SPS and PPS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730999
+
+2015-08-15 14:45:34 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Copy metadata in the payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the video tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-15 11:41:40 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-15 11:30:36 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: fix potential crash when shutting down
+ A race condition in the state change function may cause buffers to be
+ unreffed while they are still used by the streaming thread in
+ gst_rtp_h265_pay_send_vps_sps_pps() resulting in a crash. Chain up to the
+ parent class first in the state change function to make sure streaming
+ has stopped and only then free those buffers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741381
+
+2015-08-14 15:08:08 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtph265pay: fix buffer leak when using SPS/PPS
+ Fixes a buffer leak that would occur if the pipeline was shutdown while a
+ SPS/PPS header was being created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741271
+
+2015-08-14 11:49:51 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ rtph265depay: copy metadata in the depayloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the video tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-12 17:54:52 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: checking if depay has sps/pps nals before insertion
+ Related to: https://bugzilla.gnome.org/show_bug.cgi?id=753430
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 17:22:42 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: only update the srcpad caps if something else than the codec_data changed
+ h264parse and gstrtph264depay do the same, let's keep the behaviour
+ consistent. As we now include the codec_data inside the stream, this causes
+ less caps renegotiation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 16:43:48 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: PPS replaces old PPS if it has the same id
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 16:11:00 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: Insert SPS/PPS NALs into the stream
+ rtph264depay does the same and this fixes decoding of some streams with 32
+ SPS (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255),
+ but the field in the codec_data for the number of SPS or PPS is only 5
+ (or 8) bit. As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
+ This looks like a mistake in the part of the spect about the codec_data.
+
+2015-08-12 15:49:50 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't need to map the
+ input buffer again but can just re-use the mapping the base class has
+ already done.
+ Based on: https://bugzilla.gnome.org/show_bug.cgi?id=750235
+ https://bugzilla.gnome.org/show_bug.cgi?id=753228
+
+2015-08-12 15:14:50 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
+ Switching to GST_BUFFER_TIMESTAMP() to be consistent with other rtp code.
+
+2015-08-12 14:59:53 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: prevent trying to get 0 bytes from adapter
+ This causes an assertion and would lead to getting a NULL instead
+ of a buffer. Without proper checking this would easily lead to a
+ segfault.
+ Related to rpth264depay: https://bugzilla.gnome.org/show_bug.cgi?id=737199
+
+2015-07-29 17:29:28 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtp: remove dead assignment
+ Value set to ret will be overwritten at least once at the end of the while
+ loop, removing assignment.
+
+2015-04-24 16:48:23 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ remove unused enum items PROP_LAST
+ This were probably added to the enums due to cargo cult programming and are
+ unused.
+
+2015-03-06 14:54:41 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtp: donl_present variable unused
+ donl_present is not implemented, yet the value is set and checked a few times.
+ Cleaning this.
+ CID #1249687
+
+2015-01-08 15:36:04 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265pay.c:
+ rtp: value truncated too short creates dead code
+ type is truncated to 0-31 with "& 0x1f", but right after that it is checks if
+ the value is equivalent to GST_H265_NAL_VPS, GST_H265_NAL_SPS, and
+ GST_H265_NAL_PPS (which are 32, 33, and 34 respectively). Obviously, this will
+ never be True if the value is maximum 31 after the truncation.
+ The intention of the code was to truncate to 0-63.
+
+2015-01-08 15:27:44 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtp: fix nal unit type check
+ After further investigation the previous commit is wrong. The code intended to
+ check if the type is 39 or the ranges 41-44 and 48-55. Just like gsth265parse.c
+ does. Type 40 would not be complete.
+
+2015-01-08 13:47:09 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtp: fix dead code and check for impossible values
+ nal_type is the index for a GstH265NalUnitType enum. There are two types of dead
+ code here:
+ First, after checking if nal_type is >= 39 there are two OR conditionals that
+ check if the value is in ranges higher than that number, so if nal_type >= 39
+ falls in the True branch those other conditions aren't checked and if it falls
+ in the False branch and they are checked, they will always also be False. They
+ are redundant.
+ Second, the enum has a range of 0 to 40. So the checks for ranges higher than 41
+ should never be True.
+ Removing this redundant checks.
+ CID 1249684
+
+2014-10-16 10:34:01 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtp: add h265 RTP payloader + depayloader
+
+2016-02-15 11:51:46 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tests/check/elements/rtpmux.c:
+ tests: rtpmux: Fix element memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=762057
+
+2016-02-12 20:57:29 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/monoscope.c:
+ monoscope: rework the scaling code
+ The running average was wrong and the resulting scaling factor was only held in
+ place using the CLAMP. In addtion we are now convering quickly to volume
+ changes.
+ FInally now with this change, we can change the resolution defines and
+ everythign adjusts.
+
+2016-01-28 17:00:55 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/convolve.c:
+ * gst/monoscope/monoscope.c:
+ * gst/monoscope/monoscope.h:
+ monoscope: use constants in the drawing code
+ Make all the drawing ops be based on the constants. This way we can change
+ the fixed size at least at compile time.
+
+2016-01-28 09:51:17 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/gstmonoscope.c:
+ monoscope: replace hardcoded values by constants
+ This at least establishes the relationship.
+
+2016-01-28 09:43:12 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/convolve.c:
+ * gst/monoscope/convolve.h:
+ * gst/monoscope/monoscope.c:
+ * gst/monoscope/monoscope.h:
+ monoscpe: make the convolver use dynamic memory
+ Replace all #defines with members and initialize the convolver with a parameter.
+
+2016-01-28 08:56:44 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/README:
+ monoscope: update README
+ We can already create multiple instances.
+
+2016-01-28 08:53:35 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/monoscope/convolve.c:
+ * gst/monoscope/monoscope.c:
+ monoscope: code cleanup
+ Use constants more often. Cleanup comments and add more to explain how things
+ work.
+
+2016-02-08 23:41:32 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: remove check for impossible condition
+ Commit bd27a1f30b4458f2edee53c76dd07fb35904b61d added a few error handling
+ memory management checks. These check srccaps to see if it needs to be
+ unreferenced before returning, in the case of invalid_caps this goto jump
+ always happens before srccaps is set, so it will always be NULL in this
+ error label.
+ CID #1352035
+
+2016-02-08 12:48:46 +0100 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES
+ https://bugzilla.gnome.org/show_bug.cgi?id=761705
+
+2016-02-08 15:31:55 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix spelling of reenqueueing
+ To match commit 7d7074cef0272cd5155098bfc2bda6849dd89267. I love the idea
+ of aiming for the maximum number of consecutive vowels.
+
+2016-02-08 10:17:49 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix spelling of queueing
+ Didn't know which one to choose between queuing and queueing, so I picked
+ the one with the biggest amount of vowels in a row ;-P (both are
+ acceptable apparently)
+
+2016-02-07 15:02:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Don't pass the same data over and over
+ We already pass the entire frame to the decoder. If the decoder ask for
+ more data, don't pass the same data again as this leads to infinit loop.
+ Instead, simply fail the fill function to signal the problem with that
+ frame. It will then be skipped properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761670
+
+2016-02-08 00:10:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/lzo.c:
+ matroska: get rid of _stdint.h include
+
+2016-02-05 20:00:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/Makefile.am:
+ tests: extend the AM_TESTS_ENVIRONMENT from check.mak
+ To get the CK_DEFAULT_TIMEOUT defined for all tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=761472
+
+2016-02-05 18:04:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 86e4663 to b64f03f
+
+2016-01-30 18:43:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ rtpjpegpay: Skip APP and JPG markers and print warnings for unknown markers
+ For APP/JPG markers the size is following and we have to skip that. This is
+ not really a problem unless the marker contains e.g. a preview JPEG or
+ something else that we might interprete as another marker.
+
+2016-01-26 22:37:30 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix framerate calculation for fragmented format
+ qtdemux calculates framerate using duration and the number of sample.
+ In case of fragmented mp4 format, however, the number of sample can
+ be figure out after parsing every moof box. Because qtdemux does not
+ parse every moof in QTDEMUX_STATE_HEADER state, it will cause incorrect
+ framerate calculation.
+ This patch will triger gst_qtdemux_configure_stream() for every new moof.
+ Then, framerate will be calculated by using duration and n_samples of the moof.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760774
+
+2016-01-28 22:36:23 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: handling zero segment-duration edit list
+ Based on document ISO_IEC_14496-12, edit list box can have
+ segment duration as zero. It does not imply that media_start equals to
+ media_stop. But, it just indicates a sample which should be presented
+ at the first. This patch derives segment duration using media_time
+ and duration of file. And set derived duration to segment-duration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760781
+
+2016-01-28 21:36:54 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: expose streams with first moof for fragmented format
+ In case of push mode, qtdemux expose streams after got moov box.
+ We can not guarantee that a moov box has sample data such as sample duration
+ and the number of sample in stbl box for fragmented format case.
+ So, if a moov has no sample data, streams will not be exposed until get the first moof.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760779
+
+2016-01-27 18:48:17 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Check for subset instead of non-empty intersection for ACCEPT_CAPS
+
+2016-01-27 18:44:23 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Unset RECONFIGURE flag on srcpad whenever we configure new caps
+ Prevents double-negotiation during startup and in some other cases.
+
+2016-01-27 16:43:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Add negotiation unit tests for all 4 modes
+ These now check the output caps based on the input caps and a following
+ capsfilter and make sure the caps are exactly as expected.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760995
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 17:39:20 +0100 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Do passthrough in auto mode if downstream only supports interlaced
+ If the following conditions are met:
+ 1) upstream and downstream caps are compatible
+ 2) upstream is interlaced
+ 3) downstream doesn't support progressive mode
+ then deinterlace will just do passthrough instead of failing to link.
+ This is done with the following scenario in mind:
+ videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
+ name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
+ queue ! deinterlace name=dein_desktop ! autovideosink
+ In this case, dein_src will do the deinterlacing. However,
+ videotestsrc ! "video/x-raw,interlace-mode=interleaved" ! deinterlace
+ name=dein_src ! tee name=t ! queue ! deinterlace name=dein_file ! filesink t. !
+ queue ! deinterlace name=dein_desktop ! autovideosink t. ! queue !
+ "video/x-raw,interlace-mode=interleaved" ! fakesink
+ In this case, caps auto-negotiation will make dein_file and dein_desktop do
+ the deinterlacing, while dein_src will be passthrough.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760995
+
+2016-01-26 18:05:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: Add mode=auto-strict
+ In this mode we will passthrough all progressive caps but interlaced caps must be
+ caps where we actually support deinterlacing.
+ This is the only difference between auto and auto-strict, auto would
+ passthrough all unsupported interlaced caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 17:50:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Implement reconfiguration a bit better
+ And e.g. consider reconfiguration caused by RECONFIGURE events too.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+
+2016-01-26 11:57:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Rewrite caps negotiation
+ Previously the result of the CAPS query and ACCEPT_CAPS depended on what kind
+ of caps were last set, and e.g. if we last had interlaced caps or not. That's
+ just broken.
+ Also previously the handling of non-sysmem caps features was rather random and
+ unusuable.
+ Now the behaviour is the following, depending on the mode property:
+ 1) mode=disabled
+ Completely do passthrough of everything
+ 2) mode=interlaced
+ Only accept formats we can actually deinterlace, and accept interlaced
+ and progressive content and always run the deinterlacer and output
+ progressive content
+ 3) mode=auto (i.e. playbin)
+ Accept all progressive formats as passthrough, accept all formats that we
+ can deinterlace ourselves (which we do then), but also accept everything
+ else for which we then just passthrough. In auto mode, deinterlacing is best
+ effort: If we can, we deinterlace, if we can't we just output interlaced
+ content.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720388
+ https://bugzilla.gnome.org/show_bug.cgi?id=760553
+
+2016-01-26 11:34:40 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Remove unused, obsolete bufferalloc code
+
+2016-01-26 18:50:38 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: use A_AAC instead of A_AAC/MPEGx/y
+ Some GoogleCast compatible devices ignore A_AAC/MPEGx/y tracks; Also according to http://wiki.multimedia.cx/index.php?title=Matroska A_AAC/MPEGx/y is obsolete
+ https://bugzilla.gnome.org/show_bug.cgi?id=761144
+
+2016-01-25 17:21:24 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/rtp/gstrtph261pay.c:
+ gst: Fix unintialized variable warnings
+ While cross-compiling with Linaro GCC 5.1-2015.08, it complained
+ about a couple unitialized variables.
+ This patch initializes them to zero.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761094
+
+2016-01-25 15:03:23 +0100 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxsrc: print potentially negative offset with a sign
+
+2016-01-21 17:41:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Re-add colorimetry field for RGB formats
+ This time, check if it's an RGB format and sets the transformation
+ matrix to identity. The rest of the colorimetry information is
+ meaningfull and shall be kept.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759624
+
+2016-01-22 10:03:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: fix sRGB colorspace definition
+ V4l2 can also use the sRGB colorspace for YUV formats and thus needs a
+ default matrix.
+
+2016-01-21 15:29:46 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/debugutils/gsttaginject.c:
+ taginject: fix sample pipeline in docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=679571
+
+2016-01-21 10:49:44 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Add adobe colorspace support
+ Use the new primaries and transfer function for Adobe RGB.
+ Explicitly list the colorimetry instead of using the default GStreamer
+ ones. The defaults for BT2020, for example, do not match.
+ Explicitly set the matrix of SRGB to RGB.
+
+2016-01-20 13:41:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Ensure that we always have valid frame user data before using it
+ Otherwise we're going to dereference NULL pointers.
+
+2016-01-20 10:02:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: Unref frame in all code paths of handle_frame()
+ https://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-19 22:49:20 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/vpx/gstvpxenc.c:
+ vpxenc: Unref frame on ERROR
+ All code paths for handle_frame() must somehow take ownership of the frame, be
+ it by actually unreffing, forwarding the frame elsewhere or storing it for
+ later.
+ http://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-20 18:20:43 +1100 Jan Schmidt <jan@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2: Don't free props structure twice.
+ gst_v4l2_device_provider_probe_device() frees the passed props
+ structure, don't free it again in the caller.
+
+2016-01-19 15:15:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Cleanup uneeded return statement
+
+2016-01-19 15:14:59 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't set colorimetry for non YUV formats
+ Setting colormetry in caps for RGB have no meaning, but worst it
+ confuses the converters downstream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759624
+
+2016-01-19 13:01:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpchannels.c:
+ * gst/rtp/gstrtpchannels.h:
+ rtp: fix compiler warnings with gcc-6
+ In file included from gstrtpL16depay.h:27:0,
+ from gstrtp.c:73:
+ gstrtpchannels.h:154:33: error: 'channel_orders' defined but not used [-Werror=unused-const-variable]
+ static const GstRTPChannelOrder channel_orders[] =
+
+2016-01-19 14:57:03 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Don't play anything after the end of the data chunk even when seeking
+ Especially in push mode we would completely ignore the size of the data chunk
+ when not stop position is given for the seek. Instead make sure that the end
+ offset is at most the end of the data chunk if known.
+ Without this we would output anything after the data chunk, possibly causing
+ loud noises if the media file is followed by an INFO chunk or an ID3 tag.
+
+2016-01-19 14:55:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Don't do calculations with -1 offsets when handling SEGMENT events
+ We use that to signal "infinity", taking the difference between that and some
+ other value is not going to give us any useful result for the end offsets of
+ segments.
+
+2016-01-18 11:30:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
+ This reverts commit 271501f6576de4d141e7c2f618e28b9e3b1e5b38.
+ It wasn't meant to be pushed yet as the commit message indicates.
+
+2016-01-12 14:01:21 -0800 Aleix Conchillo Flaqué <aconchillo@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
+ We check the stream profile and use the proper RTCP caps:
+ application/x-srtcp if we are using a secure profile and
+ application/x-rtcp otherwise.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760556
+
+2016-01-05 16:15:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ WIP: rtpjitterbuffer: Add RFC7273 media clock handling
+
+2016-01-15 11:36:35 +0000 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/vpx/gstvpxenc.c:
+ vp8enc: Return FLOW_ERROR when an error accures
+ FALSE would mean FLOW_OK
+ https://bugzilla.gnome.org/show_bug.cgi?id=760666
+
+2016-01-15 03:57:45 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: break as soon as the device is found
+ No need to loop further if there's no side-effects for it
+
+2016-01-15 03:56:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: Fix error handling when selecting/opening devices
+ Post an element error when the CoreAudio device cannot be selected or opened.
+ Also ensure that we post a GST_ERROR with more detail.
+
+2016-01-13 23:40:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: When flushing on EOS, don't process more data than the "data" size
+ Even if we have more data queued up when flushing than the size of the data
+ chunk, don't process and output it. If the data size is known, this likely
+ contains another chunk (e.g. an INFO chunk) or things like ID3 tags. Just
+ outputting them as if they were data is going to cause unexpected behaviour
+ and unpleasant audio noises.
+
+2014-08-29 15:40:23 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * tests/check/pipelines/wavenc.c:
+ tests: fix a thinko in the wavenc example
+ The code is supposed to follow somehow what the comment above says, that
+ is to have one channel with a wave of freq 440 and the other channel
+ with a wave of freq 880, but an off by one error results in frequencies
+ of 0 and 440.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735673
+
+2014-08-29 15:07:58 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * gst/interleave/interleave.c:
+ interleave: Fix the example by setting channel-masks in the sink pads
+ The current example does not work, it fails with:
+ ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0: Internal data flow error.
+ gstwavparse.c(2178): gst_wavparse_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstWavParse:wavparse0:
+ streaming task paused, reason not-negotiated (-4)
+ This is because negotiation with wavenc gets messed up by the missing
+ channel positions configuration.
+ The proper way to define the channel layout when using the interleave
+ element in code would be to set the channel-positions property, but
+ gst-launch-1.0 does not know how to deal with arrays; so the example
+ pipeline works around the issue by setting the channel-masks in the sink
+ pads.
+ Also fix a repetition in the deinterleave example description
+ https://bugzilla.gnome.org/show_bug.cgi?id=735673
+
+2016-01-11 16:29:55 +0000 Tim Sheridan <tim.sheridan@imgtec.com>
+
+ * gst/audioparsers/gstsbcparse.c:
+ sbcparse: Fix frame length calculation
+ SBC frame length calculation wasn't being rounded up to the nearest byte
+ (as specified in the A2DP 1.0 specification, section 12.9). This could
+ cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
+ calculated frame lengths.
+ Incorrect frame length calculation causes frame coalescing to fail, as
+ subsequent frames in the stream aren't found in the expected locations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742446
+
+2016-01-10 22:54:12 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: demote warning on wrong reserved value to fixme
+ We are likely just parsing a backward-compatible stream we
+ don't fully support.
+
+2016-01-08 16:27:05 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: simplify caps selection
+ The downstream caps query with a filter alraedy gives us the possible
+ intersection so there is no need to check it again with downstream
+ if it is supported. Just try to set it directly.
+
+2016-01-07 20:42:41 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: fix unnecessary sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then
+ throw it away immediately, just use the original input buffer
+ directly.
+
+2016-01-07 20:38:27 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpdvdepay.c:
+ rtpdvdepay: fix unnecessary sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then
+ throw it away immediately, just use the original input buffer
+ directly.
+
+2016-01-07 20:34:05 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpamrdepay.c:
+ rtpamrdepay: fix unnecessary sub-buffer creation
+ We create a sub-buffer just to copy over its metas and then
+ throw it away immediately, just use the original input buffer
+ directly.
+
+2016-01-07 20:27:29 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: fix major memory leak and performance issue
+ We call gst_rtp_buffer_get_payload() which creates a sub-buffer
+ of each input buffer, just to copy over metas, and then leak it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760289
+
+2016-01-08 15:32:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/rganalysis.c:
+ rganalysis: Fix compiler warnings in the unit test
+ elements/rganalysis.c:919:66: error: shifting a negative signed value is undefined
+ [-Werror,-Wshift-negative-value]
+ push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0));
+ ~~ ^
+ elements/rganalysis.c:929:69: error: shifting a negative signed value is undefined
+ [-Werror,-Wshift-negative-value]
+ push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14));
+ ~~ ^
+ elements/rganalysis.c:939:64: error: shifting a negative signed value is undefined
+ [-Werror,-Wshift-negative-value]
+ push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14));
+ ~~ ^
+
+2016-01-05 18:13:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: don't map buffer multiple times when parsing
+
+2016-01-07 18:20:30 +0200 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: Store subtitle stream count in the correct variable
+ And don't override the video stream count instead.
+
+2016-01-05 18:59:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/equalizer/gstiirequalizernbands.c:
+ equalizer: The child-proxy API is GObject based in 1.x
+ Not GstObject anymore.
+
+2015-05-21 17:41:12 +0200 Pablo Anton <pablo.anton@vodalys-labs.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2-*: Configuring output pool correctly for using drivers min_buffer if present.
+ Signed-off-by: Pablo Anton <pablo.anton@vodalys-labs.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=755736
+
+2015-12-31 15:46:31 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: add debug msg on CRC mismatch while validating frame header
+
+2015-12-31 16:00:49 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop unneeded braces at _parse_frame() exit
+ Additionally, drop redundant comment & line break
+
+2015-12-31 15:55:18 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: minor grammar correction
+
+2015-12-31 15:34:57 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: update URLs on pointers to online spec
+
+2015-12-31 14:40:15 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: make buffer DTS setting explicitly unconditional
+ We are setting it to PTS regardless of block_strategy
+
+2015-12-31 14:21:40 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: add actual invalid block type to warning
+ For someone that read the spec is clear the only *invalid*
+ data block type is 127. For the rest, its useful information.
+ Additionally. values 7-126 are currently reserved by the
+ spec so the situation might change in the future.
+
+2015-12-31 14:12:36 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: use shift instead of mask & comp
+ We are only interested on the first bit of the first
+ byte of the metadata block header to figure out whether
+ is marked as the last one. The shift makes it quite
+ clearer.
+
+2015-12-31 12:52:13 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: warn on wishful parsing of weird headers
+ If we get anything from 7 to 126 as type when parsing
+ a metadata block header, we are likely dealing with a
+ FLAC stream version we don't fully understand. Issue
+ a warning if so.
+ Document function assumptions regarding the passed-on
+ type while at this.
+
+2015-12-31 11:33:45 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: show meaningful info on frame CRC check
+ As CRCs are calculated for the comparition already, we
+ might as well (cheaply) inform the user how the numbers
+ differ if a missmatched pair is found.
+ While at it:
+ Rephrase candidate-frame message to make more sense
+
+2015-12-31 02:40:43 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop remaining trailing whitespace
+
+2015-12-31 02:15:06 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop superflous else clauses
+
+2015-12-31 01:09:51 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: factor out buffer time and offset resetting
+ Avoids multiple occurrences of the same resetting pattern
+
+2015-12-31 00:54:48 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: move block handling by type out of _parse_frame()
+
+2015-10-07 18:51:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: replace duplicated codes to call new base sdp apis
+ https://bugzilla.gnome.org/show_bug.cgi?id=745880
+
+2015-12-30 12:16:56 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: drop redundant return statement on _header_is_valid()
+ Fix the rather vague error message while at it.
+
+2015-12-30 01:56:26 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: rework gst_flac_parse_frame_is_valid()
+ drop unnecessary nesting looking for end of frame
+
+2015-12-30 00:37:04 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: factor out context clearing routine
+
+2015-12-29 18:05:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Guard against no codec data in prores caps creation
+ CID 1346532
+
+2015-12-29 17:58:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: Initialize buffer variable to NULL
+ False positive but trivial to fix and possibly causing compiler warnings at
+ some point in the future too.
+ CID 1346535
+
+2015-07-27 15:53:26 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2deviceprovider: add properties to the device
+ Add properties to the device with exactly the same keys and sematics
+ as what pulseaudio uses as property keys.
+ Also handle the case when a device is probed manually and not through gudev.
+ https://bugzilla.gnome.org//show_bug.cgi?id=759780
+
+2015-12-25 11:41:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ scaletempo: Free the various buffers in GstBaseTransform::stop()
+ Previously we leaked them completely, but as they're specific to the caps
+ freeing them in stop() instead of finalize() makes most sense.
+
+2015-12-24 15:28:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.1 ===
+
+2015-12-24 14:16:21 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.7.1
+
+2015-12-24 13:19:24 +0100 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
+
+2015-12-24 12:22:32 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/cs.po:
+ * po/de.po:
+ * po/el.po:
+ * po/hu.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/ru.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-12-21 09:57:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: drop flushes from our own offset seek
+ Prevents downstream from receiving flushes for a seek only in
+ upstream. Those seeks are only to start reading from the right
+ offset when skipping or returning to qt atoms.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758928
+
+2015-11-11 16:53:19 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Always set the channel mask for PCM streams
+ Just use the gst_audio_channel_get_fallback_mask function for now as
+ the specification is too complicated and nobody implements it.
+
+2015-12-21 11:37:26 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Fix sleep for buffer-time lower than 200000
+ https://bugzilla.gnome.org/show_bug.cgi?id=748680
+
+2015-12-21 12:31:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Use -Bsymbolic-functions if available
+ While this is more useful for libraries, some of our plugins with multiple
+ files and some internal API can also benefit from this.
+
+2015-12-18 15:34:52 +0000 William Manley <will@williammanley.net>
+
+ * gst/debugutils/progressreport.c:
+ * gst/debugutils/progressreport.h:
+ progressreport: add support for using format=buffers with do-query=false
+ This is useful for investigating and debugging pipelines which are
+ producing buffers at a slower/faster rate than you would expect.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759635
+
+2015-12-18 15:49:43 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Update formats table
+ This change add all the new RGB based format. Those format removes the
+ ambiguity with the ALPHA channel. Some other missing multiplanar format
+ has been added with some additional cleanup.
+
+2015-12-18 05:17:15 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't write invalid edit list start time.
+ Avoid writing a negative number as a large positive
+ integer in an edit list when the first_ts is smaller
+ than the first_dts - which can happen when the first
+ packet received has a PTS but no DTS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759615
+
+2015-12-04 23:16:45 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Only update running time when it increases.
+ Don't increment running time from every buffer. The correct
+ logic to only increment when running time advances is a
+ little further down, so delete this left-over line.
+
+2015-11-18 11:01:20 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Implement prores support
+ https://bugzilla.gnome.org/show_bug.cgi?id=758258
+
+2015-11-18 16:20:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroska-demux: Play ProRes video streams
+ Generate video/x-prores caps for ProRes video streams.
+ Every frame needs an 8 byte header prepended, as described in
+ http://wiki.multimedia.cx/index.php?title=Apple_ProRes#Frame_layout
+ so do that in a post-processing callback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758258
+
+2015-12-18 10:18:09 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * ext/dv/gstdvdec.h:
+ dvdec: Remove unused fields
+ Remove unused fields frame_len and space
+ https://bugzilla.gnome.org/show_bug.cgi?id=759614
+
+2015-12-17 16:03:04 +0100 Vincent Dehors <vincent.dehors@openwide.fr>
+
+ * gst/rtp/gstrtpj2kdepay.c:
+ rtpj2kdepay: Push one JPEG2000 frame per buffer, not a buffer list with multiple buffers
+ https://bugzilla.gnome.org/show_bug.cgi?id=758943
+
+2015-12-16 11:43:58 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * ext/raw1394/gstdv1394src.c:
+ * ext/raw1394/gsthdv1394src.c:
+ dv1394: log error if failed to set socket status flag
+ Log an error message if failed to set write or read socket as
+ non-blocking.
+ CID 1139608
+ CID 1139609
+
+2015-12-15 17:10:00 +0000 Dave Craig <davecraig@unbalancedaudio.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: Check for NULL return value of gst_pad_get_current_caps()
+ https://bugzilla.gnome.org/show_bug.cgi?id=759503
+
+2015-12-16 09:35:53 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * 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 to git
+
+2015-12-15 14:27:22 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/Makefile.am:
+ vpx: Add missing headers in Makefile.am
+ This fixes distcheck.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-09-24 12:57:00 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * ext/vpx/Makefile.am:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp8enc.h:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ * ext/vpx/gstvpxenc.c:
+ * ext/vpx/gstvpxenc.h:
+ vpx: created common baseclass GstVPXEnc
+ GstVP8Enc and GstVP9Enc has almost 80% code in common.
+ created common baseclass GstVPXEnc for GstVP8Enc and GstVP9Enc
+ https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-12-15 12:57:53 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvpxdec.c:
+ * ext/vpx/gstvpxdec.h:
+ vpxdec: Remove unneeded add video_meta
+ This also remove copies for VP8, which was not correctly in place
+ in previous related patch.
+
+2015-12-15 09:49:24 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * ext/vpx/Makefile.am:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8dec.h:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9dec.h:
+ * ext/vpx/gstvpxdec.c:
+ * ext/vpx/gstvpxdec.h:
+ vpx: created common base class GstVPXdec for vpx decoders
+ Base class for the vp8dec and vp9dec.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755510
+
+2015-06-10 09:17:08 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * configure.ac:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: Add GTlsInteraction property
+ https://bugzilla.gnome.org/show_bug.cgi?id=750709
+
+2015-12-14 09:05:06 -0500 Evan Callaway <evan.callaway@ipconfigure.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Retry connection if tunneling needs authentication
+ Leverage response from gst_rtsp_connection_connect_with_response to
+ determine if the connection should be retried using authentication. If
+ so, add the appropriate authentication headers based upon the response
+ and retry the connection.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749596
+
+2015-12-14 14:19:05 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: check port-range format
+ The string could exist but with a wrong format, in that case we still want
+ to reset the values of client_port_range.min and max like we do if there is
+ no string.
+ CID 1139593
+
+2015-12-14 14:55:12 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Check device property and fail if device can't be found
+ Don't use default if a specific device is set but it can't be found.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759452
+
+2015-12-14 14:15:00 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Fix handling of the mute property
+ - set mute value at startup
+ - correct set and get mute functions
+ https://bugzilla.gnome.org/show_bug.cgi?id=755106
+
+2015-12-11 11:23:13 +0100 Thomas Roos <thomas.roos@industronic.de>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: Check the return value of GetStatus() too to decide if there was an error
+ If GetStatus() fails, the status itself won't be very meaningful but we also
+ have to look at its return value. This fixes blocking pipelines when removing
+ sound devices or during other errors, where we wouldn't notice the error and
+ then wait forever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734098
+
+2015-12-10 17:41:46 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ isomp4: remove unused parameters in build_*_extension
+ AtomTRAK parameter is not used by build_mov_alac_extension(),
+ build_jp2h_extension(), or build_mov_alac_extension() and can be
+ removed.
+
+2015-12-10 15:11:07 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: replace variable only used once
+ Replace has_shift variable with value since it is only use once.
+
+2015-12-09 12:24:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix packet dropping after a big discont
+ We would queue 5 consective packets before considering a reset and a proper
+ discont here. Instead of expecting the next output packet to have the current
+ seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're
+ going to drop all queued up packets.
+
+2015-12-09 11:49:02 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/interleave/interleave.h:
+ interleave: Remove unsed field
+ Remove unused field collect_event in interleave.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759226
+
+2015-12-07 16:33:14 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Stop pushing data as soon as possible in push-mode
+ When working in push-mode, we attempt to push out everything currently
+ buffered in the adapter.
+ This has two pitfalls:
+ * We could stop earlier (the moment we get a non-ok or non-not-linked)
+ * We return the last combined flow return, which might be completely
+ different from the previous combined flow return
+
+2015-12-07 09:08:09 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b319909 to 86e4663
+
+2015-12-07 14:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add a warning if an empty RTCP packet is tried to be sent
+ https://bugzilla.gnome.org/show_bug.cgi?id=759119
+
+2015-11-30 19:20:13 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8dec.h:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9dec.h:
+ vpxdec: Use GstMemory to avoid copies
+ With the VPX decoders it's not simple to use downstream buffer pool,
+ because we don't know the image size and alignment when buffers get
+ allocated. We can though use GstAllocator (for downstream, or the system
+ allocator) to avoid a copy before pushing if downstream supports
+ GstVideoMeta. This would still cause a copy for sink that requires
+ specialized memory and does not have a GstAllocator for that, though
+ it will greatly improve performance for sink like glimagesink and
+ cluttersink. To avoid allocating for every buffer, we also use a
+ internal buffer pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745372
+
+2015-11-30 08:42:35 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Avoid over-skipping when checking LOAS config
+ There might be multiple LOAS config in a row in a full frame. The first
+ one might be a multi-layer config (which we can't properly parse yet)...
+ but then followed by a valid (single-layer) one.
+ The code was previously skipping whole frames (instead of just the LOAS
+ config we failed to read) resulting in multiple frames (seen up to 6s in
+ some situation) being dropped before finally getting the configuration.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758826
+
+2015-11-25 17:08:56 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Properly set SPARSE stream flags for subpicture/subtitle
+ And while we're at it, also detect 'DXSA' as being a variant fourcc
+ of 'DXSB' for XSUB
+
+2015-11-30 21:23:52 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: grammar fix
+
+2015-11-30 21:01:17 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: switch shoutcast stream provider
+ Fixes failing ICY test. Previous provider has
+ streaming disabled outside UK.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758114
+
+2015-11-18 16:10:11 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
+
+ * gst/avi/gstavimux.c:
+ avimux: don't crash if we never got audio caps before stopping
+ auds.blockalign is set once the first caps arrive. If
+ gst_avi_mux_stop_file() is called before this happens then auds.blockalign
+ is zero and gst_avi_mux_audsink_set_fields() cause a crash:
+ [...]
+ avipad->parent.hdr.rate = avipad->auds.av_bps / avipad->auds.blockalign;
+ [...]
+ https://bugzilla.gnome.org/show_bug.cgi?id=758912
+
+2015-12-01 18:20:23 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: don't block when resurecting a buffer
+ When we are resurecting a buffer, don't block. instead let us copy a
+ buffer.
+
+2015-12-01 00:30:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: remove extra variable to improve readability
+ Makes it easier to see that the event is being replaced/unrefed
+
+2015-12-01 00:22:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: respect seqnum in seek events
+ Propagate the original seek seqnum to events originated from
+ seeking to make sure they have the same value
+
+2015-12-01 00:03:21 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: flush upstream when seeking in pull mode
+ Makes sure upstream will unblock and return the thread so that
+ seeking can continue
+ https://bugzilla.gnome.org/show_bug.cgi?id=758861
+
+2015-11-27 09:27:29 +0100 Anton Bondarenko <antonbo@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: add "send SPS/PPS with every key frame" mode
+ It's not enough to have timeout or event based SPS/PPS information sent
+ in RTP packets. There are some scenarios when key frames may appear
+ more frequently than once a second, in which case the minimum timeout
+ for "config-interval" of 1 second for sending SPS/PPS is not sufficient.
+ It might also be desirable in general to make sure the SPS/PPS is
+ available with every keyframe (packet loss aside), so receivers can
+ actually pick up decoding immediately from the first keyframe if
+ SPS/PPS is not signaled out of band.
+ This patch adds the possibility to send SPS/PPS with every key frame. This
+ mode can be enabled by setting "config-interval" property to -1. In this
+ case the payloader will add SPS and PPS before every key (IDR) frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-11-27 09:03:51 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtph264pay: change config-interval property type from uint to int
+ This way we can use -1 as special value, which is nicer than MAXUINT.
+ This is backwards compatible even with the GValue API, as shown by
+ a unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757892
+
+2015-11-26 21:46:11 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for Opus
+ Add support for demuxing Opus encapsulated in MP4 files, based on the
+ following spec: https://www.opus-codec.org/docs/opus_in_isobmff.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=742643
+
+2015-11-25 22:48:32 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: use macro for codec_name
+ Use _codec() macro instead of duplicating code.
+
+2015-03-25 16:32:55 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: videodec: choose format from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=733827
+
+2015-03-27 15:02:33 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: add gst_v4l2_object_probe_caps
+ Add a variant of gst_v4l2_object_get_caps that bypasses the probed_caps cache.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733827
+
+2015-11-19 17:20:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2.c:
+ v4l2-probe: Skip devices without supported formats
+
+2015-11-13 12:35:59 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ * sys/v4l2/gstv4l2.c:
+ v4l2: Track /dev/video* to triggered required probe
+ If something in /dev/video* get added, removed or replaced, we need to
+ probe the devices again in order to ensure the dynamic devices are up to
+ date.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758085
+
+2015-11-25 14:51:40 +1100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: rtpsession: don't send empty RTCP packets
+ generate_rtcp can produce empty packets when reduced size RTCP is turned on.
+ Skip them since it doesn't make sense to push them and they cause errors with
+ elements that expect RTCP packets to contain data (like srtpenc).
+
+2015-11-24 10:57:28 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: restore the segment on case of soft reset
+ When seeking back to restore the mdat position a flush is pushed
+ through and it resets downstream segment information. Make sure
+ that after the flush (that does a soft reset) a segment will
+ be pushed again
+ Fixes regressions spotted at
+ https://ci.gstreamer.net/job/GStreamer-master-validate/2100/
+
+2015-11-20 12:44:22 +0000 Graham Leggett <minfrin@sharp.fm>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: fix spelling of variable
+ https://bugzilla.gnome.org/show_bug.cgi?id=758390
+
+2015-11-20 11:05:51 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: unite duplicate FourCC
+ Unite in fourcc.h the FourCCs that are used twice or more in qtdemux
+
+2015-11-19 15:33:45 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Fix capture/output-io-mode properties
+ There was some miss-match in the implementation. This makes it
+ concistent, though functionally it worked, except the video decoder
+ output-io-mode getter.
+
+2015-11-19 19:48:06 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ atoms: remove unused argument of build_mov_wave_extension()
+ AtomTrak * trak argument of build_move_wave_extension() isn't used.
+ Removing it.
+
+2015-11-19 19:28:20 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: remove duplicate FourCC
+ Use the available FourCCs in fourcc.h instead of duplicating them.
+
+2015-11-19 18:36:39 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ isomp4: centralize all FourCC
+ 10 FourCCs generated with GST_MAKE_FOURCC() in gstqtmux.c and atoms.c
+ already exist in fourcc.h. Don't duplicate these and use them directly.
+ Plus moving 6 to fourcc.h, to centralize them all.
+
+2015-11-19 17:32:12 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/matroska/webm-mux.c:
+ matroska/webmmux: fix outdated example launch lines
+ Update gst-launch-0.10 lines to gst-launch-1.0
+
+2015-11-16 13:26:50 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ isomp4: add support for Opus in mp4mpux
+ Add support for muxing MP4 files containing Opus. Based on the spec
+ detailed here:
+ https://www.opus-codec.org/docs/opus_in_isobmff.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=742643
+
+2015-11-18 19:10:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Replace tabs with spaces
+
+2015-11-18 19:07:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Cast to signed integers to prevent unsigned compare between negative and positive numbers
+ This fixes seeking if the first entries in the samples table are negative. The
+ binary search would always fail on this as the array would not be sorted if
+ interpreting the negative numbers as huge positive numbers. This caused us to
+ always output buffers from the beginning after a seek instead of close to the
+ seek position.
+ Also add a case to the comparison function for equality.
+
+2015-11-18 16:01:48 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: remove duplicate check
+ We want 1 or 2 streamheaders, the check if (bufarr->len != 1 &&
+ bufarr->len != 2) is enough. Not need to check if bufarr->len is <= 0 or
+ > 255.
+
+2015-11-18 14:48:36 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Fix error leak and handle error
+ g_thread_try_new allows for possiblity of failures. In case it fails,
+ error is not handled and leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758260
+
+2015-11-15 17:16:29 -0800 Josep Torra <n770galaxy@gmail.com>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ rtpgstdepay: Properly handle backward compat for event deserialization
+ Actual code is checking for a NULL terminator and a ';' terminator,
+ for backward compat, in a chained way that cause all events being rejected.
+ The proper condition is to reject the events when terminator isn't
+ in ['\0', ';'] set.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758151
+
+2015-11-15 17:11:02 -0800 Josep Torra <n770galaxy@gmail.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: rtp-payloading: Test for handling of custom events in rtpgst
+ Add a simple test that checks proper serialization/deserialization
+ of custom events with rtpgstpay and rtpgstdepay.
+
+2015-11-16 16:23:43 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vpxdec: Use threads on multi-core systems
+ This adds an automatic mode to the threads property of vpxdec in order to
+ use as many threads as there is CPU on the platform. This brings back
+ GStreamer VPX decoding performance closer to what is achieved by other
+ players, including Chromium.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758195
+
+2015-11-16 10:58:32 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only send initial gaps for non-fragmented streams
+ It would be unusual to have the header segment with an 'edts' atom
+ indicating gaps at the beginning when handling fragmented streams.
+ The header usually doesn't contain any timestamping information, this
+ should come from the playlist/manifest and the segments with media
+ in those scenarios.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758171
+
+2015-11-17 09:41:34 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "Revert "qtdemux: respect qt segments in push-mode for empty starts""
+ This reverts commit d842ff288a9d01214a046becbfd9cbff3a4acea0.
+ This was reverted by accident
+
+2015-11-17 12:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: Add "loop" property for enabling/disabling multicast loopback
+ On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On Windows it
+ is a setting for the receiver socket. As such we will need it on udpsrc too to
+ allow filtering out our own multicast packets.
+
+2015-11-16 13:52:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ Revert "qtdemux: respect qt segments in push-mode for empty starts"
+ This reverts commit 142d8e2d23e5602e7382977af1043d621625f8c8.
+
+2015-11-16 16:56:04 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix string memory leak
+ The string got using g_strdup_printf will be allocated memory
+ and should be freed after use.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758161
+
+2015-11-14 21:51:11 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2/object: remove unnecessary NULL check before g_free()
+
+2015-11-14 21:45:29 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/oss/gstosssrc.c:
+ osssrc: remove unnecessary NULL check before g_free()
+
+2015-11-14 21:43:24 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * sys/sunaudio/gstsunaudiosrc.c:
+ sunaudiosrc: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:36:30 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:31:08 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: remove unnecessary NULL checks before g_free()
+
+2015-11-14 21:26:21 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska/read-common: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:43:10 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4/atoms: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:35:54 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ rtp/theorapay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:33:54 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtp/vorbispay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:31:34 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ rtp/jpegpay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:27:04 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:22:09 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:14:25 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: remove unnecessary NULL check before g_free()
+
+2015-11-14 20:09:54 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstop.c:
+ effectv/optv: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:05:03 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstshagadelic.c:
+ effectv/shagadelictv: remove unnecessary NULL checks before g_free()
+
+2015-11-14 20:01:43 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstripple.c:
+ effectv/ripple: remove unnecessary NULL checks before g_free()
+
+2015-11-14 19:56:57 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gstradioac.c:
+ effectv/radioac: remove unnecessary NULL checks before g_free()
+
+2015-11-14 19:52:12 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/effectv/gststreak.c:
+ effectv/streak: remove unnecessary NULL check before g_free()
+
+2015-11-14 17:04:55 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/shout2/gstshout2.c:
+ shout2: remove unnecessary NULL checks before g_free()
+
+2015-11-14 16:57:13 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/vpx/gstvp9enc.c:
+ vp9enc: remove unnecessary NULL check before g_free()
+
+2015-11-14 16:54:42 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: remove unnecessary NULL check before g_free()
+
+2015-11-14 16:20:33 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: remove unnecessary NULL checks before g_free()
+
+2015-11-13 13:34:02 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: add support of NV16, NV61 and NV24 formats
+ Mapped respectively to V4L2_PIX_FMT_NV16/V4L2_PIX_FMT_NV16M,
+ V4L2_PIX_FMT_NV61,V4L2_PIX_FMT_NV61M and V4L2_PIX_FMT_NV24 v4l2 formats.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758058
+
+2015-11-11 14:10:53 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmuxpartreader: Fix GCond leak
+ inactive_cond is not being cleared resulting in memory leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757924
+
+2015-08-06 12:44:20 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: fix output state memory leak
+ When jpeg_finish_decompress is called, output state reference is being created.
+ But if there is any failures in finishing decompress, it jumps to setjmp,
+ and at that point state was not referenced. Resulting in leak of output state.
+ Hence adding another setjmp after output state is referenced.
+ Similarly adding another setjmp to unmap the frame in case error happens before
+ finish_decompress
+ https://bugzilla.gnome.org/show_bug.cgi?id=753087
+
+2015-08-10 11:23:45 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: respect qt segments in push-mode for empty starts
+ In push-mode it is hard to support qt segments overall but it is
+ possible to support when the file isn't heavily edited but just contain
+ a segment to indicate a gap at the beginning. This also allows properly
+ timestamping data that has negative DTS in push-mode.
+ It is relevant to support those for 2 scenarios:
+ 1) fragmented streaming
+ 2) HTTP playback of 'regular' mp4
+ https://bugzilla.gnome.org/show_bug.cgi?id=753484
+
+2015-11-05 18:39:33 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/pulse/pulsedeviceprovider.c:
+ pulse: Don't leak caps and structures in the device provider
+
+2015-11-04 19:01:20 +0530 Arun Raghavan <arun@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: Document properties that are expressed in bits per second
+ This changed in 928cd110bcea5d143cab3ea747991851d52ecbad and
+ 73c0c2920f9aca96982a4de0c20b3417aa148b81 but was not documented.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-11-04 18:51:32 +0530 Arun Raghavan <arun@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: Trivial gst-indent fixes
+
+2015-08-12 13:35:40 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: support for cenc auxiliary info parsing outside of moof box
+ When the cenc aux info index is out of moof boundaries, keep track of
+ it and parse the beginning of the mdat box, before the first sample.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755614
+
+2015-11-03 20:33:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Use codecutils helpers for creating Opus caps
+ Also fix up codec data with values from the container.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-11-03 14:51:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: There is no multistream field for Opus anymore
+ https://bugzilla.gnome.org/show_bug.cgi?id=757152
+
+2015-11-03 12:42:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/webm-mux.c:
+ matroska/webmmux: Support Opus in webmmux and VP9 in matroskamux
+ https://bugzilla.gnome.org/show_bug.cgi?id=729950
+
+2015-11-03 12:40:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Parse and handle CodecDelay, SeekPreroll and DiscardPadding
+ https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 12:18:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Write CodecDelay, DiscardPadding and SeekPreroll for Opus
+ And also adjust timestamps and durations according to the codec delay, both
+ should include it for whatever reason.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 11:49:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Opus headers are not in-band
+ https://bugzilla.gnome.org/show_bug.cgi?id=727305
+
+2015-11-03 22:01:07 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/v4l2/gstv4l2.c:
+ v4l2: Set O_CLOEXEC on the device fd
+ This is needed to make sure that child processes don't inherit the video
+ device fd which can cause problems with some drivers.
+
+2015-11-03 14:46:30 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpmanager: switch G_GINT64_FORMAT for GST_STIME_ARGS
+ No need to use G_GINT64_FORMAT for potentially negative values of
+ GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
+ Plus it creates more readable values in the logs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757480
+
+2015-11-03 14:26:29 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff
+ No need to manually handle negative values of diff, GST_STIME_ARGS does
+ exactly this.
+
+2015-11-02 16:53:15 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: use GST_STIME_ARGS for GstClockTimeDiff
+ No need to manually handle negative values of diff, GST_STIME_ARGS does
+ exactly this.
+
+2015-11-02 16:43:46 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: use GST_STIME_ARGS for GstClockTimeDiff
+ No need to manually handle negative values of diff, GST_STIME_ARGS is
+ available for this.
+
+2015-10-30 10:05:37 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audiochebband.c:
+ audiochebband: Fix typo in example pipeline
+ Fix typo in example pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757340
+
+2015-10-28 23:47:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ v4l2: fix double-unref in the v4l2 device provider
+
+2015-10-27 10:48:00 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-ids.c:
+ matroskamux: don't drop JPEG frames that only have PTS but no DTS set
+ For the MS/VfW codec ids, we want to write DTS timestamps instead
+ of PTS because that's what everyone else seems to do (and it's also
+ how it is in AVI). So for those input formats we use the buffer DTS
+ instead of the PTS. However, if there's no DTS set but only the PTS
+ then just take the PTS instead of dropping the input buffer. This
+ is useful especially for I-frame only codecs like JPEG and huffyuv,
+ but should also be fine as fallback in general.
+ Fixes regression with input JPEG frames that only have PTS set on them.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756967
+
+2015-10-24 23:57:38 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * tests/check/elements/splitmux.c:
+ tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads
+ https://bugzilla.gnome.org/show_bug.cgi?id=753622
+
+2015-10-24 23:57:29 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: do not destroy the multiqueue & muxer when going to NULL
+ Instead, delay it until all request pads have been released. This is
+ because the release_pad() vfunc requires the multiqueue and muxer to
+ be there in order to release their request pads as well. If those
+ elements are destroyed earlier, release_pad() does not work, no
+ pads are released and some resources are leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753622
+
+2015-10-20 15:28:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Read buffer timestamp *after* actually setting it
+ https://bugzilla.gnome.org/show_bug.cgi?id=756809
+
+2015-10-24 17:14:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ * gst/audiofx/gstscaletempo.h:
+ scaletempo: Fix handling of rate < 0
+ We have to reverse all samples in a buffer before processing them to properly
+ have continuous data from one buffer to another. As a result we will have a
+ negative applied rate and a rate of 1.0.
+ Also make sure that input buffers are correctly clipped to the segment,
+ otherwise our calculations are going to go wrong.
+ Also copy over the segment event's sequence number to the output segment while
+ we're at it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757033
+
+2015-10-19 18:04:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: break as soon as non-interlaced if found
+ It looks for a non-interlaced entry on the filter caps, break
+ as soon as one is found to avoid wasting cpu
+
+2015-10-19 17:50:28 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: implement accept-caps
+ Implement accept-caps handler to avoid doing a full caps query
+ downstream to handle it.
+ This commit implements accept-caps as a simplification of the _getcaps
+ function, so it exposes the same limitations that getcaps would.
+ For example, not accepting renegotiation to caps with capsfeatures when
+ it was last configured to a caps that it has to deinterlace.
+
+2015-10-19 17:06:28 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/deinterlace.c:
+ tests: deinterlace: fix small typo in comment
+
+2015-10-26 00:41:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/files/Makefile.am:
+ check: Dist splitvideo0[012].ogg test files.
+
+2015-10-23 20:16:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ * gst/audiofx/gstscaletempo.h:
+ scaletempo: Add support for F64
+
+2015-10-22 17:40:38 -0700 Mischa Spiegelmock <mspiegelmock@gmail.com>
+
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * gst/multipart/multipartdemux.c:
+ * gst/rtp/README:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/udp/gstudpsrc.c:
+ docs: Minor fixes in various places
+ https://bugzilla.gnome.org/show_bug.cgi?id=756996
+
+2015-10-21 17:43:31 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/goom/plugin_info.c:
+ goom: remove compiler trick
+ After commit 2cb6cfed22166b262ae50cb58f3ff11dd8ba91f9 there is no need to
+ trick the compiler anymore about the usage of variable cpuFlavour.
+
+2015-10-21 14:35:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b99800a to b319909
+
+2015-10-21 17:41:38 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audiofxbaseiirfilter.h:
+ audiofx: remove unused variable
+ Remove unsued variable have_coeffs in audiofxbaseiirfilter
+ https://bugzilla.gnome.org/show_bug.cgi?id=756905
+
+2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Use new GST_ENABLE_EXTRA_CHECKS #define
+ https://bugzilla.gnome.org/show_bug.cgi?id=756870
+
+2015-10-21 14:25:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 9aed1d7 to b99800a
+
+2015-10-21 11:53:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: relax creation time parsing
+ Parse wrong timestamps like we used to write as well,
+ e.g. 10:9:42, and the hour might be without a leading
+ zero in any case.
+
+2015-10-21 11:45:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix indentation
+
+2015-10-21 11:44:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: extract both creation date and time
+ Before we only extracted the date part.
+
+2015-10-21 11:16:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: fix writing of creation time
+ Don't write time as e.g. 11:9:42
+
+2015-10-13 12:42:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/rtp/gstrtpj2kpay.c:
+ rtpj2kpay: update fragment offset
+ It was always being set to 0, making the resulting stream broken
+ for the receiver
+ https://bugzilla.gnome.org/show_bug.cgi?id=756422
+
+2015-10-19 15:36:37 +0300 Ryan Hendrickson <ryan.hendrickson@alum.mit.edu>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't unconditionally use strnlen()
+ It's not available on older OSX and we can as well use memchr() here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756154
+
+2015-10-19 17:38:32 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/auparse/gstauparse.c:
+ auparse: Fix event memory leak
+ Free the event after being handled to prevent memory leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756799
+
+2015-10-19 09:14:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: unify raw audio caps into a single caps structure
+
+2015-10-14 15:42:50 -0700 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for FFV1 coded streams in mov
+ https://bugzilla.gnome.org/show_bug.cgi?id=752495
+
+2015-10-14 15:53:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: EOS immediately if we have an empty seek segment
+ https://bugzilla.gnome.org/show_bug.cgi?id=748316
+
+2015-10-14 10:43:19 +0300 Stavros Vagionitis <stavrosv@digisoft.tv>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Make non-inclusive segment boundaries inclusive
+ The problem is that the filesrc and souphttpsrc are behaving
+ differently regarding the calculation of the segment boundaries. The
+ filesrc is using a non-inclusive boundaries, while the souphttpsrc
+ uses inclusive. Currently the hlsdemux calculates the boundaries as
+ inclusive, so for this reason there is no problem with the souphttpsrc,
+ but there is an issue in the filesrc.
+ The GstSegment is non-inclusive, so the proposed solution is to use
+ non-inclusive boundaries in the hlsdemux in order to be consistent.
+ Make the change in the hlsdemux, will break the souphttpsrc, which
+ will expect inclusive boundaries, but the hlsdemux will offer
+ non-inclusive. This change makes sure that the non-inclusive
+ boundaries are converted to inclusive.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748316
+
+2015-10-11 22:07:54 +0000 Graham Leggett <minfrin@sharp.fm>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpclientsink.h:
+ souphttpclientsink: Add the retry and retry-delay properties
+ These allow a failed request to be retried after the given number of seconds
+ instead of failing the pipeline. Take account of the Retry-After header if
+ present. Add retries parameter that controls the number of times an HTTP
+ request will be retried before failing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756318
+
+2015-10-14 12:03:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix caps leak
+ If the QtDemuxStream are re-used they may already have caps which used
+ to be leaked.
+ Reproduced using the
+ validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
+ scenario.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756561
+
+2015-10-14 09:29:50 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix taglist memory leak
+ Free the stream and its sub items instead of just the stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=756544
+
+2015-10-11 12:06:26 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Allow negotiating to S8 as a raw format but stop making it best choice
+ Negotiation to audio/x-raw,format=S8 was not possible because S8 does
+ not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
+ https://bugzilla.gnome.org/show_bug.cgi?id=756387
+
+2015-10-11 09:18:40 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Add prores support
+ https://bugzilla.gnome.org/show_bug.cgi?id=756388
+
+2015-10-12 18:56:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: add GST_PLUGINS_BASE_LIBS for flvdemux check
+ So it pulls in the right libgsttag-1.0.
+
+2015-10-11 22:27:47 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * gst/goom/Makefile.am:
+ * gst/goom/gstaudiovisualizer.c:
+ * gst/goom/gstaudiovisualizer.h:
+ * gst/goom/gstgoom.h:
+ * gst/goom2k1/Makefile.am:
+ * gst/goom2k1/gstaudiovisualizer.c:
+ * gst/goom2k1/gstaudiovisualizer.h:
+ * gst/goom2k1/gstgoom.h:
+ goom/goom2k1: remove obsolete left over files
+ They now use the new GstAudioVisualizer base class
+ from gst-plugins-base/gst-libs/gst/pbutils
+ Also fixed undefined reference to gst_audio_visualizer_get_type
+ Added GST_PLUGINS_BASE_LIBS to Makefile.am and re-order LIBADD.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-12 10:48:23 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ mpegaudioparse: Fix buffer memory leak during failures
+ mapped buffer is not being unmapped during failures
+ https://bugzilla.gnome.org/show_bug.cgi?id=756231
+
+2015-10-12 11:18:51 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Check if soup message is created
+ If soup message is not created then the same should not be passed
+ on, which is resulting in segfault. Hence throwing a warning message
+ and returning
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:15:15 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Check if location being set is valid
+ Adding a check in set_property to find if the location uri is valid
+ and printing warning if not valid.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:09:30 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Fix memory leaks during failures
+ freeing streamheader_buffers and sent_buffers during failure cases.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-12 11:03:17 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Replace redundant free_buffer_list function
+ Removing free_buffer_list and replacing it with already available function
+ g_list_free_full
+ https://bugzilla.gnome.org/show_bug.cgi?id=755326
+
+2015-10-11 16:40:01 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/Makefile.am:
+ check: Don't forget base CFLAGS for flvdemux check
+ elements/flvdemux.c:25:25: fatal error: gst/tag/tag.h: No such file or directory
+
+2015-10-11 11:37:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/ebml-write.c:
+ * gst/matroska/ebml-write.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroskamux: Create a TIME segment when creating streamable output
+ Related to https://bugzilla.gnome.org/show_bug.cgi?id=754435 which
+ does the same for flvmux.
+
+2015-09-23 13:50:52 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/Makefile.am:
+ * gst/flv/gstflvdemux.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/flvdemux.c:
+ flvdemux: output speex vorbiscomment as a GstTagList
+ This is what speexdec expects.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755478
+
+2015-09-22 22:59:16 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ * tests/check/elements/flvmux.c:
+ flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
+ Or else flvdemux don't understand it
+ https://bugzilla.gnome.org/show_bug.cgi?id=754435
+
+2015-09-02 10:44:59 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ * tests/check/elements/flvmux.c:
+ flvmux: use time segment and copy timestamps when streamable
+ Add a basic test using speex data to verify timestamping.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754435
+
+2015-09-23 13:14:03 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: speex is also always 16KHz
+ This is just a cosmetic change for the logs, since the right caps
+ for Speex is being set elsewhere.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755479
+
+2015-07-14 15:19:44 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpmanager: Add 'source-stats' to stats and notify
+ Add statitics from each rtp source to the rtp session property.
+ 'source-stats' is a GValueArray where each element is a GstStructure of
+ stats for one rtp source.
+ The availability of new stats is signaled via g_object_notify.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752669
+
+2015-06-05 17:20:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Implement sending of reduced size RTCP packets
+ https://bugzilla.gnome.org/show_bug.cgi?id=750456
+
+2015-10-08 15:01:13 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audiodynamic.h:
+ audiofx: Remove unused variable
+ Remove unused variable 'degree' in audiodynamic
+ https://bugzilla.gnome.org/show_bug.cgi?id=756234
+
+2015-10-08 14:44:07 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix memory leak for corrupted file
+ Free brands before overriding them.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756226
+
+2015-10-08 11:44:04 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ gdkpixbufdec: Fix pixbuf_loader leak during failures
+ https://bugzilla.gnome.org/show_bug.cgi?id=756219
+
+2015-10-07 23:23:45 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Add missing break
+
+2015-10-07 13:03:02 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2015-10-07 13:02:12 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpmanager: add "max-dropout-time" and "max-misorder-time" props
+ https://bugzilla.gnome.org/show_bug.cgi?id=751311
+
+2015-10-07 17:14:57 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix date memory leak
+ When getting date from taglist, the memory should be freed after
+ using it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756171
+
+2015-10-05 11:03:38 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix sample memory leak
+ When getting sample from taglist, the memory should be freed after
+ using it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756068
+
+2015-10-05 13:10:56 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/cutter/gstcutter.c:
+ cutter: Fix buffer leak
+ Buffer is added to the internal cache, and pushed only when accumulated
+ buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
+ is not freed. Freeing the cache when the state changes from PAUSED to READY.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754212
+
+2015-08-31 21:10:16 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Use default upstream event handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-08-31 21:05:03 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ rtpmux: As 0xFFFFFFFF is a valid ssrc, check if it has been set
+ https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-07-22 09:47:22 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ * tests/check/elements/rtpmux.c:
+ gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers
+ By not doing this, the muxer is not effectively a rtpmuxer, rather a
+ funnel, since it should be a single stream that exists the muxer.
+ If not specified, take the first ssrc seen on a sinkpad, allowing upstream
+ to decide ssrc in "passthrough" with only one sinkpad.
+ Also, let downstream ssrc overrule internal configured one
+ We hence has the following order for determining the ssrc used by
+ rtpmux:
+ 0. Suggestion from GstRTPCollision event
+ 1. Downstream caps
+ 2. ssrc-Property
+ 3. (First) upstream caps containing ssrc
+ 4. Randomly generated
+ https://bugzilla.gnome.org/show_bug.cgi?id=752694
+
+2015-10-02 22:42:20 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fixup last commit
+
+2015-10-02 22:21:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * gst/udp/gstudpsrc.c:
+ Update GLib dependency to 2.40.0
+
+2015-06-30 16:56:19 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpstats.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpstats: add utility for calculating RTP packet rate
+
+2015-08-10 18:14:39 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: handle empty segments in seeking adjust
+ If seeking targets an empty segment skip it as there is no media
+ offset to get from it. Instead look for the next one.
+ This doesn't make seeking in push-mode work if you seek to an
+ empty segment but at least won't get you to wrong offsets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753484
+
+2015-04-17 14:25:43 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: post messages when fragments are being opened and closed
+ This can be useful for applications that need to track the created fragments
+ (to log them in a recording database, for example)
+ https://bugzilla.gnome.org/show_bug.cgi?id=750108
+
+2015-04-29 18:23:28 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: allow non-video streams to serve as reference
+ In the absence of a video stream, the first stream will be used as
+ reference.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753617
+
+2015-07-22 17:45:12 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: initialize mux_start_time properly
+ mux_start_time refers to the running_time of the buffer
+ that goes first in the output file. Normally this time is
+ 0, so this variable is initialized to 0 during the state
+ change to PAUSED.
+ However, when dealing with dynamic pipelines and starting
+ a recording while the pipeline has already run for a while,
+ the running_time of the first buffer is > 0 and this causes
+ a problem with detecting the end of the first file(s) when
+ splitting by duration, because the code will later compare
+ the threshold_time with (last buffer running_time - mux_start_time)
+ and will get it wrong until mux_start_time advances enough
+ to make this difference < threshold_time, creating empty files
+ in the meantime.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753624
+
+2015-09-16 16:03:02 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Reverse playback does not consider segment.start
+ During reverse playback, the media should stop playing at segment.start
+ This does not happen, and avidemux continues to process data even when
+ current timestamp is less that segment.start.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755094
+
+2015-09-23 12:39:35 +0900 Manasa Athreya <manasa.athreya@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check multi trex to find track id in mp4 mpeg-dash stream
+ If stream has more than one trex box which is not matched to actual
+ track id, it makes qtdemux crashed.
+ Author : Manasa Athreya (manasa.athreya@lge.com)
+ https://bugzilla.gnome.org/show_bug.cgi?id=754864
+
+2015-09-04 14:24:45 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: get size, stride info using VideoInfo
+ Use VideoInfo data to get size stride and
+ offset, instead of hard coded macros.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754558
+
+2015-09-04 14:18:50 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: free mask
+ Free the memory allocated to 'mask' to avoid
+ memory leak.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754555
+
+2015-08-20 11:02:58 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ * tests/icles/equalizer-test.c:
+ * tests/icles/gdkpixbufoverlay-test.c:
+ * tests/icles/gdkpixbufsink-test.c:
+ * tests/icles/test-oss4.c:
+ * tests/icles/videocrop-test.c:
+ gstreamer: good: tests: Fix memory leaks when context parse fails.
+ When g_option_context_parse fails, context and error variables are not getting free'd
+ which results in memory leaks. Free'ing the same.
+ And replacing g_error_free with g_clear_error, which checks if the error being passed
+ https://bugzilla.gnome.org/show_bug.cgi?id=753853
+
+2015-10-02 16:18:15 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: doesn't handle probation and rtp gap in case of sender
+ https://bugzilla.gnome.org/show_bug.cgi?id=754548
+
+2015-10-02 16:16:32 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpsession.h:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpmanager: add new on-new-sender-ssrc, on-sender-ssrc-active signals
+ Allows for applications to get internal source's RTP statistics.
+ (eg. sender sources for a server/client)
+ https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-10-02 14:17:48 +1000 Jan Schmidt <jan@centricular.com>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Gather and coalesce all damaged areas before retrieving.
+ These days the xserver seems to give us the same damage regions
+ over and over for entire windows, and we retrieve them multiple
+ times, which gives time for more damage to appear. Instead, just
+ quickly gather all damaged areas into a region list and copy
+ out once.
+
+2015-10-01 16:24:32 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/goom2k1/Makefile.am:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: use the new audiovisualizer base class
+ Rebase to have goom using the GstAudioVisualizer base class in
+ gst-plugins-base/gst-libs/gst/pbutils
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-10-01 16:16:08 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/goom/Makefile.am:
+ * gst/goom/gstgoom.h:
+ goom: use the new audiovisualizer base class
+ Rebase to have goom using the GstAudioVisualizer base class in
+ gst-plugins-base/gst-libs/gst/pbutils
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-09-30 17:35:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ * tests/check/elements/deinterleave.c:
+ deinterleave: implement accept-caps
+ Avoid using default accept-caps handler that will query downstream
+ and is more expensive. Just check if the caps is compatible with
+ the template and check if the channels are the same.
+
+2015-09-30 09:35:39 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/deinterleave.c:
+ tests: deinterleave: also check for caps query results
+
+2015-09-30 12:30:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: use the caps query filter
+ It was being ignored and would lead to wrong results if the
+ element doing the query would rely on the intersection being made.
+
+2015-09-30 10:00:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: implement a caps query handler for the sinkpad
+ It was missing and apparently code relied on having it there
+ for not allowing a change in the number of channels
+
+2015-09-30 09:05:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: fix caps leak
+ Caps from the pad template are being leaked. In any case it is
+ from a static pad template and will 'leak' in the end, just doing
+ the cleanup for the good practice.
+
+2015-09-29 11:15:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/gdkpixbufoverlay.c:
+ tests: gdkpixbufoverlay: add minimal unit test
+ https://bugzilla.gnome.org/show_bug.cgi?id=755773
+
+2015-09-29 11:12:48 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ gdkpixbufsink: don't leak old pixel buffer when setting a new overlay
+ https://bugzilla.gnome.org/show_bug.cgi?id=755773
+
+2015-09-28 20:25:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: avoid potential string overflow
+ We don't necessarily have full control over the input tags, so
+ it's possible that the ISRC tag contains a longer string than
+ expected, in which case we'd write over the end of the static-size
+ 13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc.
+ Make sure to only copy the ISRC if it's not too long, and make
+ sure the buffer we write to is always NUL-terminated by using
+ g_strlcpy().
+ CID 1324931.
+
+2015-09-28 18:03:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Remove leftover assertion from 0.10
+ We now allocate memory via GstAllocator and as such can handle arbitrary
+ alignments, not only <= G_MEM_ALIGN.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755708
+
+2015-09-25 10:01:37 +0200 Guillaume Marquebielle <guillaume.marquebielle@parrot.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: fix uninitialized variables in LOAS config reading
+ On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
+ "Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
+ 'channels' are pointing to uninitialized values can end on setting random values as
+ rate and channels on src caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755611
+
+2015-09-18 00:58:23 +1000 Jan Schmidt <thaytan@noraisin.net>
+
+ * ext/gdk_pixbuf/gstgdkpixbufsink.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ Fix some compiler warnings when building with G_DISABLE_ASSERT
+ Touches rtpmanager and gdkpixbufsink
+
+2015-08-18 14:30:57 +0100 Chris Bass <floobleflam@gmail.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: support timed-text subtitle tracks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752818
+
+2015-09-26 00:12:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ gst: Don't use deprecated gst_segment_to_position()
+
+2015-09-21 13:47:21 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
+ https://bugzilla.gnome.org/show_bug.cgi?id=755125
+
+2015-09-16 19:28:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpbin/session: Allow RTCP sync to happen based on capture time or send time
+ Send time is the previous behaviour and the default, but there are use cases
+ where you want to synchronize based on the capture time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755125
+
+2015-09-25 23:51:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.6.0 ===
+
+2015-09-25 23:15:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.6.0
+
+2015-09-25 22:57:34 +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
+
+2015-09-25 14:08:09 +0200 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/smpte/gstsmptealpha.c:
+ smptealpha: Do not set width/height before comparing with old values
+ Otherwise we end up considering the values did not change and we wrongly
+ work with the old video format (which will lead to wrong
+ behaviour/segfaults).
+ https://bugzilla.gnome.org/show_bug.cgi?id=755621
+
+2015-09-23 20:59:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Accumulate segments for edit lists before activating the next segment
+ eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always
+ accumulating segments manually when activating a segment. This is only
+ needed when handling edit lists, not when activating a segment because of a
+ seek. Do the accumulation when switching edit list segments instead.
+ This fixes segment seeks again, while keeping edit lists playback working.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755471
+
+2015-09-23 17:43:51 +0530 Vikram Fugro <vikram.fugro@gmail.com>
+
+ * gst/spectrum/gstspectrum.c:
+ spectrum: send phase values in the GstMessage for Phase info
+ https://bugzilla.gnome.org/show_bug.cgi?id=755463
+
+2015-09-22 00:46:01 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Don't output a warning on MONO multiview mode.
+
+2015-09-19 17:02:18 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtptheoradepay.c:
+ rtptheoradepay: Fix memory leaks
+ The same memory leaks were fixed in identical fashion for
+ vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
+
+2015-09-19 17:04:07 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
+ * use g_list_free_full(), don't iterate elements maually when freeing
+ * call gst_rtp_*_pay_clear_packet(), don't duplicate its code
+ * use gst_buffer_unref() to clarify that it is buffers being released,
+ instead of refering directly to gst_mini_object_unref()
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277
+
+2015-09-19 18:44:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=755265
+
+=== release 1.5.91 ===
+
+2015-09-18 19:33:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * 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:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.5.91
+
+2015-09-18 19:23:57 +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
+
+2015-09-18 11:50:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-09-17 10:50:01 +0900 Eunhae Choi <eunhae1.choi@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Fix taglist leak
+ gst_tag_list_insert() does not take ownership of the inserted taglist.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755138
+
+2015-09-16 07:05:36 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Skip LOAS AAC until a valid config is seen.
+ It's normal when dropping into the middle of a stream to
+ not always have the config available immediately, so skip LOAS
+ until a valid config is seen without either setting invalid
+ caps or erroring out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751386
+
+2015-09-13 15:41:38 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: reset just a bit more upon flush_stop
+
+2015-09-13 15:40:09 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: remove dead struct member
+
+2015-09-11 17:09:28 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: fix GError memory leak when hostname resolution fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=754869
+
+2015-09-10 15:26:54 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/matroska/ebml-write.c:
+ matroskamux: drop HEADER flag from output buffers
+ Drop HEADER flag from output buffers if they are not indeed
+ headers.
+ Fixes resending of headers in tcp connection handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=754768
+
+2015-09-10 16:00:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/ebml-write.c:
+ matroskamux: fix matroskamux ! matroskademux
+ Don't carry over DISCONT flags from the input buffers to the
+ output buffer, or the demuxer might reset its state when it
+ receives the first data buffer just after parsing the simple
+ block header, and then expect sane data to follow.
+ Fixes matroskamux ! demux erroring out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754768
+ https://bugzilla.gnome.org/show_bug.cgi?id=657805
+
+2015-09-09 12:51:40 -0700 Martin Kelly <martin@surround.io>
+
+ * gst/rtsp/README:
+ rtsp: fix small README typo
+ https://bugzilla.gnome.org/show_bug.cgi?id=754807
+
+2015-09-04 19:45:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstwavpackparse.c:
+ wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
+ https://bugzilla.gnome.org/show_bug.cgi?id=752106
+
+2015-09-04 19:34:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
+ flac contains the sample offset in the frame header, so after a seek
+ without index flacparse will know the exact position we landed on and
+ timestamp buffers accordingly. It only set the pts though, which means
+ the baseparse-set dts which was set to the seek position prevails, and
+ since the seek was based on an estimate, there's likely a discrepancy
+ between where we wanted to land and where we did land, so from here on
+ that dts/pts difference will be maintained, with dts possibly multiple
+ seconds ahead of pts, which is just wrong. The easiest way to fix this
+ is to just set both pts and dts based on the sample offset, but perhaps
+ parsed audio should just not have dts set at all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752106
+
+2015-09-06 16:33:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ docs: remove properties and signals that no longer exist
+ https://bugzilla.gnome.org/show_bug.cgi?id=726443
+
+2013-10-11 15:13:00 +0000 George Chriss <gschriss@gmail.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Make the element count in arrays not include end
+ One-line removal of tags_written++
+ This should fix rtmp output to crtmpserver, and hopefully
+ noone is expecting that the element count includes the end
+ element, as different bits of documentation say different
+ things about whether it should or not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=661624
+
+2015-07-30 00:59:15 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Store incoming bitrate tags and send in the metadata
+ Apparently the Microsoft Azure RTMP server requires that the
+ videodatarate and audiodatarate metadata be provided, so
+ set those, even if it's to 0. Use the actual input bitrate
+ tags if available.
+
+2015-09-04 00:06:29 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't parse key data more than needed.
+ When an auxilliary streams are present in the SDP media,
+ there's no need to re-parse the SDP attributes multiple
+ times.
+
+2015-09-03 20:56:55 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
+ In parse_keymgmt(), don't mutate the input string that's been passed
+ as const, especially since we might need the original value again if
+ the same key info applies to multiple streams (RTX, for example).
+ When a resource is 404, and we have auth info - retry with the auth
+ info the same as if we had receive unauthorised, in case the resource
+ isn't even visible until credentials are supplied.
+ Fix a memory leak handling Mikey data.
+ When generating a random keystring, don't overrun the 30 byte
+ buffer by generating 32 bytes into it.
+
+2015-09-04 15:18:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix build with GLib < 2.44
+ G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.
+
+2015-09-04 12:01:52 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
+ This happens on Windows if we use the same socket for sending packets,
+ and the remote sends ICMP port/host unreachable messages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754534
+
+2015-09-02 21:12:41 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ rtpvorbis/theoradepay: Fix handling of fragmented packets
+ This was broken in b1089fb520 by not considering the full packet length of a
+ fragmented packet but only the length of the first one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754417
+
+2015-09-01 15:39:22 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/dtmf/gstdtmfsrc.c:
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ dtmfsrc: Reply to latency query
+
+2015-08-31 16:42:30 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/elements/rtpsession.c:
+ tests: Fix rtpsession test failure
+ The time of the first RTCP packet is semi-random, so
+ sometimes it was produced before enough packets from
+ the second SSRC were received. First drop queued RTCP
+ packets, then advance the clock enough to ensure
+ that at least one new RTCP packet is produced.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750731
+
+2015-08-31 13:56:04 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * tests/check/elements/level.c:
+ level: improve the test for multi-channel mode
+ Change the test to verify the read-index for multiple messages per buffer.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=754144
+
+2015-08-31 12:46:52 +0200 Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Align raw video frames to 32 bytes
+ Outputting unaligned video frames causes videoscale et al to
+ crash when attempting SIMD-accelerated conversion.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736965
+
+2015-08-26 23:16:46 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/level/gstlevel.c:
+ level: fix level calculations for mutliple channels
+ This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab.
+
+2015-08-27 10:28:55 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: Fix memory leak
+ In gst_smpte_collected(), check upfront if input formats are same
+ or not. This avoids allocation of in1 and in2 buffers and
+ subsequent memory leak when input formats do not match.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754153
+
+2015-08-21 11:52:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ tests: souphttpsrc: don't try to connect to dead radio server
+
+2015-08-21 16:29:16 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Trivial fix to check correct condition
+ When checking for describe method, because of missing parentheses, wrong
+ condition is being checked, which will result in wrong behavior.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753912
+
+2015-08-21 13:19:02 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: read: fix tag list memory leak
+ gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
+ be unref'ed as soon as the tags are merged to the tocentry
+ https://bugzilla.gnome.org/show_bug.cgi?id=753904
+
+2015-08-21 12:20:59 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/wavpack/gstwavpackdec.c:
+ wavpackdec: fix taglist memory leak
+ When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
+ by audiodecoder and the caller should free the taglist being passed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753903
+
+2015-08-20 14:45:33 +0200 Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: fix pad closing
+ Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@veo-labs.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=753875
+
+=== release 1.5.90 ===
+
+2015-08-19 13:29:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * 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:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.5.90
+
+2015-08-19 12:47:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
+
+2015-08-19 11:29:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-08-13 17:29:58 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesrc.c:
+ multifilesrc: fix regression with starting from index set via index property
+ When we haven't started yet, set the start_index when we set the index property,
+ so that we start at the right index position after the initial seek. The index
+ property was never really meant to be for writing, but it used to work, so let's
+ support it for backwards compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739472
+
+2015-08-18 10:52:11 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix offset calculation when parsing CENC aux info
+ Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for
+ DASH common encryption, however commit
+ bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master
+ shortly before the CENC commit caused the calculation of the CENC
+ aux info offset to be incorrect.
+ The base_offset was being added if present, but if the base_offset
+ is relative to the start of the moof, the offset was being added twice.
+ The correct approach is to calculate the offset from the start of the
+ moof and use that offset when parsing the CENC aux info.
+
+2015-08-17 14:28:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: actually return true for accept-caps query handling
+
+2015-08-17 14:07:10 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ rtp: copy metadata in the (de)payloaders which is missed before
+ https://bugzilla.gnome.org/show_bug.cgi?id=753706
+
+2015-08-16 15:21:51 -0400 Dustin Spicuzza <dustin@virtualroadside.com>
+
+ * configure.ac:
+ * sys/directsound/gstdirectsoundsink.c:
+ * sys/directsound/gstdirectsoundsink.h:
+ directsoundsink: allow specifying audio playback device
+ https://bugzilla.gnome.org/show_bug.cgi?id=753670
+
+2015-08-16 13:51:47 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: remove single entry if from loop
+ Iterate from the 2nd channel on and create the 1 channel struct
+ outside to make loop structure simpler and only slightly faster.
+
+2015-08-16 13:21:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: implement proper accept-caps
+ Should just compare with what can be immediatelly accepted by
+ the element. flacenc can't renegotiate so if it has a caps already
+ it should only accept if it is that caps otherwise just use the
+ template caps
+
+2015-08-16 13:03:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: improve sink pad template caps
+ Removes the need for custom caps query handling and makes it more
+ correct from the beginning on the template. It is a bit uglier
+ to read because there is 1 entry per channel but makes code easier
+ to maintain.
+
+2015-08-16 12:41:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/y4m/gsty4mencode.c:
+ y4mencode: fix gst-launch version in documentation
+
+2015-08-15 22:32:21 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/speex/gstspeexenc.c:
+ * ext/wavpack/gstwavpackenc.c:
+ * gst/law/alaw-encode.c:
+ * gst/law/mulaw-encode.c:
+ audioencoders: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+ Elements: speexenc, wavpackenc, mulawenc, alawenc
+
+2015-08-15 22:29:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/jpeg/gstjpegenc.c:
+ * ext/libpng/gstpngenc.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ * gst/y4m/gsty4mencode.c:
+ videoencoders: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+ Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc
+
+2015-08-16 17:21:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ mpegaudioparse: use new baseparse API to fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-03-17 17:50:37 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: use new base parse API to fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 14:37:53 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: use new baseparse API and fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 13:04:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use signed integer type to be able to check for negative subtraction results
+ CID 1315829
+
+2015-08-16 11:50:34 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtpvorbisdepay.c:
+ rtpvorbisdepay: remove dead code
+ payload_buffer must be NULL in ignore_reserved. Check will always be false.
+ Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf
+ CID #1316476
+
+2015-08-15 22:45:53 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-encode.c:
+ * gst/law/alaw-encode.h:
+ alawenc: port to AudioEncoder base class
+
+2015-08-15 09:16:23 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacdec.c:
+ * ext/speex/gstspeexdec.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * gst/law/alaw-decode.c:
+ * gst/law/mulaw-decode.c:
+ audiodecoders: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+ Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
+
+2015-08-15 08:49:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ * ext/libpng/gstpngdec.c:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ videodecoders: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+ Elements: jpegdec, pngdec, vp8dec, vp9dec
+
+2015-08-15 11:31:04 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-decode.c:
+ alawdec: make error handling a bit nicer
+ Print the element along with the debug to make it easier to trace
+ the failures
+
+2015-08-15 11:04:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-decode.c:
+ * gst/law/alaw-decode.h:
+ alawdec: port to audiodecoder base class
+ mulawdec was already ported, alawdec was left behind.
+
+2015-08-15 10:34:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only look for more samples in moofs in pull-mode
+ For playback of some fragmented formats with qtdemux it will
+ try to look for the next moof after finishing one but it is only
+ possible for pull-mode. For playback of streaming fragmented formats
+ such as DASH it should just not try to look for another moof but
+ instead wait for more data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752602
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-08-15 12:58:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstdcaparse.c:
+ dcaparse: Don't look for a second syncword
+ There are streams out there that consistently contain garbage between
+ every frame so we never ever find a second consecutive syncword.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=738237
+
+2015-08-15 11:12:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp8enc, vp9enc: reset multipass file index when stopping encoder
+ Fixes multipass encoding when re-using the same element/pipeline
+ for subsequent encoding runs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-15 11:09:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp9enc: provide support for multiple pass cache files
+ Some files may provide different caps insight of one stream. Since
+ vp9enc support caps reinit, we should support cache reinit too.
+ If more then file cache file will be created, the naming will be:
+ cache cache.1 cache.2 ...
+ Based on patch by: Oleksij Rempel <linux@rempel-privat.de>
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-14 11:41:42 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/aacparse.c:
+ tests: aacparse: use caps query instead of accept-caps
+ The accept-caps query just does a shallow check at the current
+ element while at this test we want it to also look at downstream.
+ So use caps query there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 11:40:22 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: enable accept-template flag
+ Do a quick check with the pad template caps as it is enough. Users
+ should have figured the appropriate full caps on a previous caps query
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 15:46:53 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: send the User-Agent header
+ Sometimes it is useful to know this information on the
+ server side. Other popular implementations (vlc, ffmpeg, ...)
+ also send this header on every message.
+ This includes a new "user-agent" property that the user
+ can set to use a custom User-Agent string. The default
+ is "GStreamer/<version>"
+ https://bugzilla.gnome.org/show_bug.cgi?id=750101
+
+2015-08-14 15:42:42 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: wrap gst_rtsp_message_init_request in a local function
+ This will allow adding common request initialization, like the
+ user agent string, in just one place.
+
+2015-08-14 09:36:09 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: make sure buffer gets reallocated if max_delay changes
+ https://bugzilla.gnome.org/show_bug.cgi?id=753490
+
+2015-07-09 09:51:26 +0200 Oleksij Rempel <linux@rempel-privat.de>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp8enc.h:
+ vp8enc: provide support for multiple pass cache files
+ Some files may provide different caps insight of one stream. Since vp8enc
+ support caps reinit, we should support cache reinit too.
+ If more then file cache file will be created, the naming will be:
+ cache
+ cache.1
+ cache.2
+ ...
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-04-15 22:51:51 +0200 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/rtp/gstrtpmp4gdepay.c:
+ rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
+ Use constantDuration to calculate the timestamp of non-first AU in the
+ RTP packet.
+ If constantDuration is not present in the MIME parameters, its value
+ must be calculated based on the timing information from two consecutive
+ RTP packets with AU-Index equal to 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747881
+
+2015-08-14 06:43:13 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: remove unnecessary if, g_free is null safe
+
+2015-08-14 08:33:56 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: add property to set HTTP method
+ To allow souphttpsrc to be use HTTP methods other than GET
+ (e.g. HEAD), add a "method" property that is a string. If this
+ property is not set, GET is used.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752413
+
+2015-08-14 11:13:01 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/generic/states.c:
+ check: Rename states unit test
+ Makes it easier to differentiate from other modules states unit test
+
+2015-08-14 09:21:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstaudiovisualizer.c:
+ * gst/goom/gstaudiovisualizer.h:
+ * gst/goom2k1/gstaudiovisualizer.c:
+ * gst/goom2k1/gstaudiovisualizer.h:
+ goom: Rename get_type() function of base class to prevent symbol conflicts
+ This is a problem when statically linking.
+
+2015-08-13 16:32:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
+ Otherwise we will just output buffers without timestamps after a reset if no
+ timestamps are provided by upstream, e.g. when using RTSP over TCP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-08-12 17:16:01 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/matroska/matroska-demux.h:
+ * gst/matroska/matroska-parse.h:
+ matroska: Remove unused variable
+ https://bugzilla.gnome.org/show_bug.cgi?id=753556
+
+2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg726depay.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/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegpay.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/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtptheorapay.h:
+ * gst/rtp/gstrtputils.c:
+ * gst/rtp/gstrtputils.h:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvorbispay.h:
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawpay.c:
+ rtp: Copy metadata in the (de)payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the video tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-10 18:20:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix small typo in comment
+
+2015-08-10 16:19:18 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/goom2k1/gstgoom.c:
+ goom2k1/doc: Fixup previous commit
+
+2015-08-10 15:55:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1/doc: Use GstGoom2k1 namespace
+ The doc generator isn't happy when we have class name clash. Simply
+ use it's own namespace.
+
+2015-08-10 17:10:42 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: removed unused variable in set_property
+ unused local variable 'delay' is removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753450
+
+2015-08-10 12:45:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix suboptimal queue iteration code
+
+2015-08-09 17:25:45 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: don't use glib 2.44-only API
+
+2015-07-29 14:14:50 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: add support for ISOBMFF Common Encryption
+ This commit adds support for ISOBMFF Common Encryption (cenc), as
+ defined in ISO/IEC 23001-7. It uses a GstProtection event to
+ pass the contents of PSSH boxes to downstream decryptor elements
+ and attached GstProtectionMeta to each sample.
+ https://bugzilla.gnome.org/show_bug.cgi?id=705991
+
+2015-08-10 14:13:50 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: checking if depay has sps/pps nals before insertion
+ https://bugzilla.gnome.org/show_bug.cgi?id=753430
+
+2015-08-08 16:44:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: fix outdated comment
+ The default behaviour was changed in the 0.10 -> 1.x
+ transition, but the comment was not updated.
+
+2015-08-08 17:42:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ rtptheorapay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:41:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:34:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtptheorapay.h:
+ rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
+ We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
+ and 4:4:4 formats.
+
+2015-08-06 17:46:13 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ rtpklv(de)pay: add "RTP" in the klass string
+ GstRTSPMedia uses this classification to detect the real payloader
+ inside a dynpay bin and asserts if it doesn't find it, therefore
+ it is required
+ https://bugzilla.gnome.org/show_bug.cgi?id=753325
+
+2015-08-05 11:13:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: use a dynamic pt in the test
+ 1) Tests that using dynamic PT instead of the default ones work
+ 2) If we ever decide to change the codec here we don't need to
+ worry about change the PT for the default one of the new codec
+ in the test
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-05 10:53:15 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: print valid type where guint32 is expected
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-06 11:33:37 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmupay.c:
+ rtppayload: set standard payload type as default
+ Initialize the PT to the default value of the codec and check if
+ it is still the default before declaring the pt to be dynamic or
+ not when setting the caps.
+ Also use the PT constants from the rtp lib when possible
+ https://bugzilla.gnome.org/show_bug.cgi?id=747965
+
+2015-07-26 12:07:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: store the moof-offset also for push mode
+ It will be used in some cases for getting the correct offsets
+ from trun atoms.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-26 02:09:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.h:
+ qtdemux: handle default-base-is-moof flag
+ Handle the flag from the tfhd that signals the base offset to
+ start from the moof atom
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-29 18:54:35 -0600 Glen Diener <grd@loganmill.net>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroskademux: Preserve forward referenced track tags
+ https://bugzilla.gnome.org/show_bug.cgi?id=752850
+
+2015-08-04 18:07:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: fix test failure
+ The RTP PT for alaw is 8.
+ Less than 50 packets are received in the length of this test so it
+ would never drop a buffer or would drop only the last buffer and
+ it would fail sometimes when the received wouldn't receive the
+ retransmission packet in time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpstreamdepay.c:
+ rtpstreamdepay: Only allow activation in push mode
+ We need a proper caps event from upstream with the full RTP caps as we can't
+ create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
+ a filesrc or any other element that supports pull mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753066
+
+2015-08-04 16:28:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ soup: fix typo in translated string
+ https://bugzilla.gnome.org/show_bug.cgi?id=753240
+
+2015-08-04 12:25:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Put the profile and level into the caps
+
+2015-08-04 12:09:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Only update the srcpad caps if something else than the codec_data changed
+ h264parse does the same, let's keep the behaviour consistent. As we now
+ include the codec_data inside the stream too here, this causes less caps
+ renegotiation.
+
+2015-08-04 11:48:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
+ The spec says:
+ When a picture parameter set NAL unit with a particular value of
+ pic_parameter_set_id is received, its content replaces the content of the
+ previous picture parameter set NAL unit, in decoding order, with the same
+ value of pic_parameter_set_id (when a previous picture parameter set NAL unit
+ with the same value of pic_parameter_set_id was present in the bitstream).
+
+2015-08-03 13:45:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: remove extra \n at debug message
+
+2015-08-03 13:42:20 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: prevent deadlock when states change too fast
+ If the GOP is completed, pads have to start gathering for the
+ next one but it is possible that the the state might go to
+ COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
+ thread has a chance to wake up and proceed, leaving it trapped in
+ the check_completed_gop loop and deadlocking the other threads
+ waiting for it to advance.
+ To solve it, this patch also checks that tha input running time
+ hasn't changed to prevent this scenario.
+
+2015-08-03 17:55:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Insert SPS/PPS NALs into the stream
+ h264parse does the same and this fixes decoding of some streams with 32 SPS
+ (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
+ the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
+ As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
+ This looks like a mistake in the part of the spec about the codec_data.
+
+2015-07-30 11:29:27 +0900 Eunhae Choi <eunhae1.choi@samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: handle empty http proxy string
+ 1) If the system http_proxy environment variable is not set
+ or set to an empty string, we must not set proxy to avoid
+ http connection error.
+ 2) In case of proxy property setting, if user want to clear
+ the proxy setting, they should be able to set it to NULL or
+ an empty string again, so this is fixed too.
+ 3) Check if the proxy string was parsed correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752866
+
+2015-07-29 15:46:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: remove unused variable
+ Remove unused variable 'framecount' from dvdemux
+ https://bugzilla.gnome.org/show_bug.cgi?id=753008
+
+2015-07-30 15:32:09 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: assertion error due to wrong condition check
+ In media to caps function, reserved_keys array is being used for variable i,
+ leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+ changed it to variable j
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-30 15:21:20 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtp/gstrtpmp4vdepay.c:
+ rtpmp4vdepay: rtpbuffer is being unref'ed twice
+ process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
+ the refernce should not be removed here
+ https://bugzilla.gnome.org/show_bug.cgi?id=753042
+
+2015-07-29 11:26:46 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Strip keys from the fmtp that we use internally in our caps
+ Skip keys from the fmtp, which we already use ourselves for the
+ caps. Some software is adding random things like clock-rate into
+ the fmtp, and we would otherwise here set a string-typed clock-rate
+ in the caps... and thus fail to create valid RTP caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 19:28:33 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Support mpegtsmux as a muxer.
+ As a fallback, look for a pad template sink_%d on
+ the muxer when requesting pads, to support mpegtsmux
+ https://bugzilla.gnome.org/show_bug.cgi?id=752999
+
+2015-06-25 01:35:27 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ splitmuxsrc: Use a separate lock to delay typefind.
+ Don't hold the main splitmux part lock over
+ the parent state change function, as it prevents
+ posting error messages that happen. Since the purpose
+ is to prevent typefinding from proceeding, use a
+ separate mutex just for that.
+
+2015-07-29 13:43:50 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: fix memory leak
+ After adding to tag list, key_val is not being free'd
+ resulting in memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=752992
+
+2015-07-27 13:34:14 +0900 Manasa Athreya <manasa.athreya@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc
+ 'NONE' and 'raw ' fourcc don't always contain U8 audio, it can
+ be more bits as well, in which case it's just like 'twos'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752613
+
+2015-07-24 15:10:05 +0200 Dimitrios Katsaros <patcherwork@gmail.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: Allow framerate to be large then 100pfs
+ This limit was arbitrary. We still fixate near 100pfs for compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752825
+
+2015-07-25 03:25:28 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Stop without posting error on flushing
+ This could just be a normal pipeline shutdown.
+
+2015-07-23 15:00:08 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also
+ https://bugzilla.gnome.org/show_bug.cgi?id=752618
+
+2015-07-16 18:09:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/matroskademux.c:
+ tests: add minmal matroskademux test for subtitle output
+ Some of the subtitle chunks will have embedded
+ NUL-terminators (last three), some don't (first three),
+ some will have markup, some won't, some will be valid
+ UTF-8 (all but last), some won't (last stanza).
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-16 18:49:26 +0300 Dimitrios Christidis <dchristidis@mykolab.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix for subtitle buffers with NUL terminators
+ Commit 45892ec8 created a regression where g_utf8_validate() would fail
+ if the subtitle buffer had a NUL terminator as part of the data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-21 13:31:05 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpvp8depay: Check available bytes before copy
+ Need to check that the number of bytes we want to copy from the adapter
+ actually is available and handle the error case gracefully. This error
+ may happen if malformed packets are received and we don't have a
+ complete frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752663
+
+2015-07-16 09:32:36 +0900 Paul Hyunil <paul.hyunil@lge.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Support subtitle when track subtype is fourcc_subt
+ https://bugzilla.gnome.org/show_bug.cgi?id=752655
+
+2015-07-20 16:59:40 +0800 Song Bing <b06498@freescale.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Set timestamp when queue buffer.
+ Should set timestamp when queue buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752618
+
+2015-07-16 15:12:17 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * tests/check/elements/rtpmux.c:
+ rtpmux: handle different ssrc's on sinkpads
+ Do this by not putting the ssrc from the src pads in the caps used to
+ probe other sinkpads, and then intersecting with it later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752491
+
+2015-07-16 17:19:03 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/avi/gstavimux.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/webm-mux.c:
+ Update mailing list address from sourceforge to freedesktop
+
+2015-07-15 13:44:52 +0300 Dimitrios Christidis <dchristidis@mykolab.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix trailing '*' displayed with some text subtitles
+ The subtitle buffer we push out should not include a NUL terminator
+ as part of the data, we just add such a terminator for safety, but
+ it should not be included in the buffer size.
+ A NUL terminator is not valid UTF-8, so checks will fail if it's
+ included in the size, and the NUL will be replaced by the fallback
+ character specified when converting, i.e. '*'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-15 18:23:05 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * ext/pulse/pulsedeviceprovider.c:
+ * ext/pulse/pulseutil.c:
+ * ext/pulse/pulseutil.h:
+ pulse: add properties to GstDevice
+ Add the extra properties we get from pulse to the GstDevice we expose
+ with the device monitor
+
+2015-07-15 17:20:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audioinvert.c:
+ * gst/audiofx/audiowsincband.c:
+ audiofx: Fix typo in example pipelines
+ Fix typo in example pipelines of audiowsincband and audioinvert.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752416
+
+2015-04-15 18:27:04 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: add a "format-location" signal that allows better control over filenames
+ In certain applications, splitting into files named after a base
+ location template and an incremental sequence number is not enough.
+ This signal gives more fine-grained control to the application to
+ decide how to name the files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750106
+
+2015-04-15 20:13:27 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudiosrc: no resampling on OS X
+ Unlike Remote IO, AUHAL doesn't have built-in resampling
+ for sources -- confirmed by Core Audio engineer Doug Wyatt:
+ http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-04-15 18:29:14 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudiosrc: avoid get_channel_layout
+ This only produces a warning and serves no purpose.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-04-07 15:40:14 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Avoid making a duplicate structure in caps for mono/stereo case
+ For 1ch or 2ch devices, we just need to set the caps to allow both
+ options since CoreAudio will up/downmix appropriately.
+ Also fixes the condition for the 2ch case to be exact, rather than at
+ least 2 channels since the downmix will not take place in the >stereo
+ case.
+
+2015-04-06 16:22:34 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Don't set the format on an initialized AudioUnit
+ We need to initialize the AudioUnit early to be able to probe the
+ underlying device, but according to the AudioUnitInitialize() and
+ AudioUnitUninitialize() documentation, format changes should be done
+ while the AudioUnit is uninitialized. So we explicitly uninitialize the
+ AudioUnit during a format change and reinitialize it when we're done.
+
+2015-04-06 15:55:59 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Minor spelling fix (unitialize -> uninitialize)
+
+2015-03-21 20:34:25 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Fix lockup in _audio_unit_property_listener
+ _audio_unit_property_listener is called either from a Core Audio thread
+ or as a result of a Core Audio API (e.g. AudioUnitInitialize)
+ from our own thread. In the latter case, osxbuf can be already locked
+ (GStreamer's mutex is not recursive).
+ We introduce the flag cached_caps_valid and use it instead of nullifying
+ cached_caps when we cannot lock on osxbuf.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-12 12:15:12 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Invalidate cached caps on format change
+ Listen for changes in hardware stream format and channel layout, and
+ invalidate cached caps (since they contain the preferred caps).
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-09 23:34:06 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosink.h:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxaudiosrc.h:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Overhaul of probing caps
+ - Probing caps is unified between source and sink
+ - Hardware stream format is now reported as preferred capabilities
+ (dynamically updated when hardware configuration changes)
+ - Get hardware channel layout from Remote IO just like from HAL
+ - More comprehensive mapping between AudioChannelLabel and
+ GstAudioChannelPosition
+ - Support for unpositioned channel layouts
+ - Announce stereo-mono upmixing/downmixing in caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-09 23:15:56 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: AudioUnitInitialize on open
+ Call AudioUnitInitialize upon open. Otherwise, we cannot get
+ (hardware) stream format nor channel layout from the outer scope.
+
+2015-07-12 14:27:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmparobustdepay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvp8depay.c:
+ rtp: depayloaders: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't
+ need to map the input buffer again but can just re-use
+ the mapping the base class has already done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-05-27 19:19:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't
+ need to map the input buffer again but can just re-use
+ the map the base class has already done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-07-10 00:13:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix indention
+
+2015-07-09 23:59:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Always estimate DTS from the current clock time
+ Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
+ we would produce wrong DTS. As now the estimated DTS is based on the clock,
+ don't store it in the jitterbuffer items as it would otherwise be used in the
+ skew calculations and would influence the results. We only really need the DTS
+ for timer calculations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-09 09:26:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/.gitignore:
+ gitignore: ignore rtph263 test
+
+2015-07-08 23:47:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2
+ Replace static constants with macros to make gcc happy
+ CC elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
+ elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
+ static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
+ ^
+ elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
+ static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
+ ^
+ elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
+ PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;
+
+2015-07-08 23:40:45 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: run indent and fix some comments
+ Fix indent on this file and break some comment lines into two to make
+ it fit 80 chars per line
+
+2015-07-08 15:02:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: rework segment event handling for adaptive streaming
+ When a new time segment is received upstream is going to restart
+ with a new atom. Make the neededbytes and todrop variables
+ reflect that to avoid waiting too much or dropping the
+ initial bytes that contain the header.
+
+2015-07-08 12:35:55 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: push data from adapter before starting new segment
+ The adapter might have data remaining from the previous segment,
+ push it all before clearing the adapter and starting a new segment.
+ It can accumulate data if it had pushed and got not-linked, returning
+ immediately without processing all the data. Before starting a new
+ segment this data should be handled.
+
+2015-07-08 19:59:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 21:08:36 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix gap-time calculation and remove "late"
+ The amount of time that is completely expired and not worth waiting for,
+ is the duration of the packets in the gap (gap * duration) - the
+ latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
+ that we make a "multi-lost" packet for.
+ The "late" concept made some sense in 0.10 as it reflected that a buffer
+ coming in had not been waited for at all, but had a timestamp that was
+ outside the jitterbuffer to wait for. With the rewrite of the waiting
+ (timeout) mechanism in 1.0, this no longer makes any sense, and the
+ variable no longer reflects anything meaningful (num > 0 is useless,
+ the duration is what matters)
+ Fixed up the tests that had been slightly modified in 1.0 to allow faulty
+ behavior to sneak in, and port some of them to use GstHarness.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738363
+
+2015-06-30 11:21:31 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
+ This reverts commit 05bd708fc5e881390fe839803b53144393d95ab0.
+ The reverted patch is wrong and introduces a regression because there
+ may still be time to receive some of the packets included in the gap
+ if they are reordered.
+
+2015-07-07 23:53:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: flush samples before adding more from moof
+ Avoids accumulating all samples from a fragmented stream that could
+ lead to a 'index-too-big' error once it goes over 50MB of data. It
+ could reach that before 2h of playback so it doesn't take that long.
+ As upstream elements are providing data in time format they should
+ be the ones that have more information about the full media index
+ and should be able to seek if possible.
+
+2015-07-07 23:56:12 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: rename upstream_newsegment to upstream_format_is_time
+ upstream_newsegment isn't really clear on what it means, it is set
+ to TRUE when the upstream element sends a segment in TIME format, so
+ rename it to be more clear about it.
+ It is important to know this because it means that upstream has
+ a notion of time and qtdemux is likely being driven by an upstream
+ element that is reading from a higher level abstraction than a file,
+ such as a DASH, MSS or DLNA element.
+
+2015-07-07 21:31:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix leak by flushing previous sample info from trak
+ In fragmented streaming, multiple moov/moof will be parsed and their
+ previously stored samples array might leak when new values are parsed.
+ The parse_trak and callees won't free the previously stored values
+ before parsing the new ones.
+ In step-by-step, this is what happens:
+ 1) initial moov is parsed, traks as well, streams are created. The
+ trak doesn't contain samples because they are in the moof's trun
+ boxes. n_samples is set to 0 while parsing the trak and the samples
+ array is still NULL.
+ 2) moofs are parsed, and their trun boxes will increase n_samples and
+ create/extend the samples array
+ 3) At some point a new moov might be sent (bitrate switching, for example)
+ and parsing the trak will overwrite n_samples with the values from
+ this trak. If the n_samples is set to 0 qtdemux will assume that
+ the samples array is NULL and will leak it when a new one is
+ created for the subsequent moofs.
+ This patch makes qtdemux properly free previous sample data before
+ creating new ones and adds an assert to catch future occurrences of
+ this issue when the code changes.
+
+2015-07-07 16:46:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix index size check and debug message
+ It is allocating samples_count + n_samples, not only n_samples
+
+2015-07-08 17:02:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Calculate receive time if we don't have any
+ This is required to properly schedule packet loss timers and make
+ sure all our calculations work properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 15:13:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations
+ That is, handle DTS==GST_CLOCK_TIME_NONE correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 20:31:42 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix event leak
+ when seek fails in avidemux, event is not being freed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752117
+
+2015-07-08 12:02:22 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtph263.c:
+ rtph263depay: Make sure payload is large enough
+ Plus new unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752112
+
+2015-07-08 08:59:49 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ rtpklvdepay: fix printf format compiler warning
+ v_len is of type guint64, but while print the value(16 + len_size + v_len)
+ G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT
+ https://bugzilla.gnome.org/show_bug.cgi?id=752100
+
+2015-07-07 20:25:47 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ docs: add new RTP elements to docs
+
+2015-07-07 20:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: rtp-payloading: add basic unit test for KLV payloading
+ Also make it so that the mtu is always set if specified, not
+ only in case of the rather weird bufferlist test code path.
+ This allows us to easily make the payloader fragment a payload
+ across multiple output packets by setting a small MTU on it.
+
+2015-07-07 19:58:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvdepay.h:
+ rtpklvdepay: improve start detection and handle fragmented KLV units
+
+2015-07-05 20:25:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvdepay.h:
+ rtp: add SMPTE 336M KLV metadata depayloader
+ http://tools.ietf.org/html/rfc6597
+
+2014-08-09 10:08:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpklvpay.c:
+ * gst/rtp/gstrtpklvpay.h:
+ rtp: add SMPTE 336M KLV metadata payloader
+ http://tools.ietf.org/html/rfc6597
+
+2015-07-07 16:59:20 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/atomsrecovery.c:
+ * gst/isomp4/properties.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/rtpmanager/rtpsource.c:
+ docs: fix "Symbol name not found at the start of the comment block"
+ Add symbols or change comment into a regular comment.
+
+2015-07-07 16:58:53 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/audioparsers/gstamrparse.h:
+ docs: remove outdated doc strings
+
+2015-07-03 23:10:40 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ docs: add missing plugins and ensure master doc is sorted
+
+2015-07-07 15:54:41 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ Revert "imagefreeze: Remove impossible error condition"
+ This reverts commit d46631c5c7312ad613397f8238c7a9714ae3ae94.
+ pad only handle EOS events but not EOS flow, and will push the buffer again
+ resulting in an assertion error. So we should not handle the buffer
+ and return EOS flow.
+
+2015-07-07 15:50:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpg729depay.c:
+ rtpg729depay: unmap rtp buffer in error path
+
+2015-07-07 15:48:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: fix buffer leak
+ The handle_buffer vfunc takes ownership of the input buffer.
+ Fixes elements/rtp-payloading under valgrind.
+
+2015-07-02 08:52:43 +0200 Tobias Mueller <muelli@cryptobitch.de>
+
+ * gst/goom/goom_core.c:
+ goom: Initialised variables to remove compiler warnings
+ goom_core.c: In function 'goom_update':
+ goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
+ ^
+ goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
+ ^
+ https://bugzilla.gnome.org/show_bug.cgi?id=752053
+
+2015-07-07 09:18:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: fix indentation
+
+2015-07-06 19:11:00 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Fix uninitialized variable compiler error
+ endpos variable does not correctly understand in the
+ 4.6.3 GCC version. So compile error appears when we do
+ compile rtph261pay using jhbuild.
+ This patch is fixed the compile error in 4.6.3 GCC version.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751985
+
+2014-11-12 12:08:58 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Handle seek flags properly
+ Allows for non-keyframe seeks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738570
+
+2015-02-24 10:50:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: avoid looping reading the 'moof' atom forever
+ It gets stuck if it only finds a moof and no mfra/mfro or moov
+ atoms. Skip the moof to continue the parsing to have it either
+ play or error out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745089
+
+2015-06-26 13:24:17 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: improve error handling
+ for files which have corrupted header, libflac is not able to
+ process the metadata properly. We just try to ignore the error
+ and continue with the processing, since metadata parsing is not
+ making much of a difference to libflac
+ https://bugzilla.gnome.org/show_bug.cgi?id=751334
+
+2015-07-06 20:16:38 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * sys/ximage/ximageutil.c:
+ ximagesrc: add meta transform function
+ ximage metadata can't be transformed or copied, but provide an empty
+ transformation function instead of NULL to allow unconditional calling
+ of metas' transform functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751778
+
+2014-06-16 16:14:28 +0200 Stian Selnes <stian.selnes@gmail.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ rtph263pdepay: init debug category
+ https://bugzilla.gnome.org/show_bug.cgi?id=752012
+
+2014-06-20 10:59:14 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpv8depay: ignore reserved bit in payload descriptor
+ Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:
+ R: Bit reserved for future use. MUST be set to zero and MUST be
+ ignored by the receiver.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751929
+
+2015-07-04 20:56:42 +0200 Stian Selnes <stian@pexip.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: rtph261depay: Add documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=751982
+
+2015-07-03 21:58:14 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-07-03 14:29:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Fix compiler warning
+ gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
+ gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
+ GObjectClass *gobject_class;
+
+2015-07-03 14:03:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261depay.c:
+ rtph261depay: Let the base class push the buffer so it can deal with the flow return
+
+2015-07-03 14:11:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Remove unused adapter
+
+2015-07-03 13:17:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpspeexpay.c:
+ speexpay: Directly attach payload to the output buffer instead of copying it
+
+2015-07-03 13:07:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpsbcpay.c:
+ sbcpay: Attach payload directly to the output instead of copying
+
+2014-12-01 14:18:40 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261depay.h:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph261pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtp: add H.261 RTP payloader and depayloader
+ Implementation according to RFC 4587.
+ Payloader create fragments on MB boundaries in order to match MTU size
+ the best it can. Some decoders/depayloaders in the wild are very strict
+ about receiving a continuous bit-stream (e.g. no no-op bits between
+ frames), so the payloader will shift the compressed bit-stream of a
+ frame to align with the last significant bit of the previous frame.
+ Depayloader does not try to be fancy in case of packet loss. It simply
+ drops all packets for a frame if there is a loss, keeping it simple.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751886
+
+2015-07-03 12:18:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpvdepay.c:
+ rtpmpvdepay: Don't forget to unmap the input buffer
+
+2015-07-03 12:14:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpvpay.c:
+ rtpmpvpay: Create buffer lists instead of pushing each buffer individually
+
+2015-07-03 12:03:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpapay.c:
+ rtpmpapay: Use buffer lists instead of pushing each fragment individually
+
+2015-07-03 10:51:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmp4apay.c:
+ rtpmp4apay: Create buffer lists and don't copy payload memory
+
+2015-06-29 16:14:18 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT
+ When there are a lot of small gaps, we can consider that there is
+ a big gap (too losses) to reset the buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751636
+
+2015-06-29 15:53:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: If possible, always update the current time before looping over all timers
+ If we have a clock, update "now" now with the very latest running time we have.
+ If timers are unscheduled below we otherwise wouldn't update now (it's only updated
+ when timers expire), and also for the very first loop iteration now would otherwise
+ always be 0.
+ Also the time is used for the timeout functions, e.g. to calculate any times
+ for the next timeouts and we would otherwise pass too old times there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751636
+
+2015-07-02 14:34:57 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: fix memory leak
+ tmp needs to be freed before going out of scope in 'done'.
+ CID #1308954
+
+2015-07-02 12:23:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263ppay.c:
+ rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it
+
+2015-07-02 09:48:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us
+
+2015-07-02 09:17:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pay.h:
+ rtph263pay: Stop using an adapter and directly use the buffer
+ We always pushed one buffer into the adapter, then handled exactly that one
+ buffer and flushed it from the adapter. Now also don't memcpy() the actual
+ payload but just attach the input buffer's data to the output buffer.
+ This code still needs some serious refactoring/rewriting.
+
+2015-07-01 21:57:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgsmpay.c:
+ rtpgsmpay: Remove non-existing includes for now
+ git add -p mistake.
+
+2015-07-01 19:29:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: Use the return value of gst_buffer_append()
+
+2015-07-01 19:19:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgsmpay.c:
+ rtpgsmpay: Attach payload to the output buffer instead of copying it
+
+2015-07-01 17:58:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: Attach payload directly to output buffers instead of copying
+
+2015-07-01 17:43:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpg723pay.c:
+ rtpg723pay: Attach payload buffer to the output instead of copying
+
+2015-07-01 17:30:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpdvdepay.c:
+ rtpdvdepay: Map the output buffer once instead of once every 80 bytes
+
+2015-07-01 21:46:46 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix return type of index_entry_offset_search()
+ It's a compare function and may return a negative value,
+ so should for correctness and consistency return a signed
+ integer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751780
+
+2015-07-01 14:12:57 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: refactor handle_next_buffer
+ The goal of this patch is making handle_next_buffer function
+ more readable avoiding unnecesary gotos and adding other
+ cosmetic changes.
+
+2015-07-01 15:40:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpac3pay.c:
+ rtpac3pay: Attach the payload to the output buffer instead of copying it
+ Might also want to produce buffer lists here if needed.
+
+2015-07-01 15:38:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ rtp: Fix indention
+
+2015-07-01 12:37:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/rtp/Makefile.am:
+ * tests/examples/rtp/client-VP8-OPUS.sh:
+ * tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh:
+ rtp: Add examples with VTS/ATS for VP8/OPUS
+ Let's have an example with modern codecs.
+
+2015-06-30 18:11:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
+
+2015-06-30 14:06:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ vp8depay: Don't lock/map every non-keyframe buffer twice
+ Just copy the complete header instead of first looking at the first byte
+ and then at the remaining 10 bytes.
+
+2015-06-29 16:05:44 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: document fallthrough cases
+ Pacify coverity and document fallthrough cases in switch statements.
+ CID #1308948, #1308947, #1308946
+
+2015-06-29 10:36:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout"
+ This reverts commit 0c21cd7177ea883c710999147ddcedb19004d182.
+ If we have multiple immediate timers, we want to first handle the one with the
+ lowest sequence number... which would be broken now.
+ Instead of this we should just use a GSequence for the timers, and have them
+ sorted first by timestamp, and for equal timestamps by sequence number. Then
+ we would always only have to take the very first timer from the list and never
+ have to look at any others.
+
+2015-06-29 10:14:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout
+ If we have lots of such immediate timeouts, we would otherwise have quadratic
+ runtime in the number of timeouts.
+
+2015-06-19 18:01:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: sticky events are sent automatically from the pad
+ No need to send them explicitly from the element
+ https://bugzilla.gnome.org/show_bug.cgi?id=751240
+
+2015-06-19 18:00:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: make sure to push sticky events before adding pad
+ It allows the caps to be set on the pad before being added for
+ dynamic autoplugging to work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751240
+
+2015-06-26 00:05:29 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property
+ Enable to use new ntp-time-source property of rtpbin
+ https://bugzilla.gnome.org/show_bug.cgi?id=751496
+
+2015-06-25 23:19:58 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpbin/session: fix description
+ https://bugzilla.gnome.org/show_bug.cgi?id=751496
+
+2015-06-25 10:57:25 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/matroska/matroska-demux.c:
+ * tests/examples/shapewipe/shapewipe-example.c:
+ docs: decodebin2 -> decodebin
+
+2015-06-25 10:47:06 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: update example pipeline
+ Update reference to decodebin2 to decodebin
+
+2015-06-25 10:45:35 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: remove dead assignments
+ Values in fields_required and same_buffer are overwritten before used. Removing
+ assignment
+
+2015-06-25 10:06:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/Makefile.am:
+ * ext/mikmod/Makefile.am:
+ * ext/mikmod/README:
+ * ext/mikmod/drv_gst.c:
+ * ext/mikmod/gstmikmod.c:
+ * ext/mikmod/gstmikmod.h:
+ * ext/mikmod/mikmod_reader.c:
+ * ext/mikmod/mikmod_types.c:
+ * ext/mikmod/mikmod_types.h:
+ * m4/Makefile.am:
+ * m4/libmikmod.m4:
+ * win32/MANIFEST:
+ * win32/vs8/libgstmikmod.vcproj:
+ mikmod: remove ancient unported plugin
+ This hasn't been touched in 11 years, and
+ clearly no one's been missing it.
+
+2015-06-23 20:15:13 +0900 Gilbok Lee <gilbok.lee@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: does not detect orientation
+ Most files don't contain the values for transposing the coordinates
+ back to the positive quadrant so qtdemux was ignoring the rotation
+ tag. To be able to properly handle those files qtdemux will also ignore
+ the transposing values to only detect the rotation using the values
+ abde from the transformation matrix:
+ [a b c]
+ [d e f]
+ [g h i]
+ https://bugzilla.gnome.org/show_bug.cgi?id=738681
+
+2015-06-25 00:04:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
=== release 1.5.2 ===
-2015-06-24 Sebastian Dröge <slomo@coaxion.net>
+2015-06-24 23:30:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.5.2
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.5.2
2015-06-24 22:56:12 +0200 Sebastian Dröge <sebastian@centricular.com>
Original commit message from CVS:
add some files
+2001-12-17 18:37:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
+
+ building up speed
+ Original commit message from CVS:
+ building up speed
+
projects / platform / upstream / gstreamer.git / blobdiff