+=== release 1.19.2 ===
+
+2021-09-23 01:33:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-good.doap:
+ * meson.build:
+ Release 1.19.2
+
+2021-09-22 14:03:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ rtph263pdepay: flag keyframes on output buffers
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1091>
+
+2021-08-18 19:47:40 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtphdrext-twcc.c:
+ rtphdrhext-twcc: Return failure on map failure
+ This feels like exactly like a case that should fail.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>
+
+2021-08-18 19:46:25 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtphdrext-colorspace.c:
+ * gst/rtpmanager/gstrtphdrext-rfc6464.c:
+ * gst/rtpmanager/gstrtphdrext-twcc.c:
+ rtphdrext: Update write() API to return a signed value
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1059>
+
+2021-09-14 17:26:27 +0900 Seungha Yang <seungha@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Try to build AAC codec-data whenever it's possible
+ AAC codec_data is a just collection of AAC profile, samplerate, and
+ channels. We can know samplerate and channels from parsed
+ SampleEntry data. Although the AAC profile is unknown there,
+ let's assume it as AAC-LC like we've been doing for the version 1
+ atom.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1082>
+
+2021-09-07 22:23:01 +0200 Vivienne Watermeier <vwatermeier@igalia.com>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ flv: fix seqnum handling for seeks
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1078>
+
+2021-01-18 16:06:27 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: also allow muxing different h264/5 profiles/levels/etc
+ All of that is advertised through the codec_data itself so can change
+ just fine within isomp4.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1071>
+
+2021-09-11 09:24:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: Add support for muxing/demuxing ffv1
+ Previously only demuxing when stored via the RIFF/AVI mapping was
+ supported.
+ See https://github.com/FFmpeg/FFV1/blob/master/ffv1.md#matroska-file-format
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/923
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1080>
+
+2021-09-12 12:18:32 +0100 Philippe Normand <philn@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: Update cache
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1081>
+
+2021-08-03 19:12:11 +0900 Seungha Yang <seungha@centricular.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Fix crash when interlaced field height is not DCT block size aligned
+ In case of interlaced JPEG file, we are doubling stride.
+ The scratch scan line should take account of it as well.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1042>
+
+2021-09-02 08:38:54 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Also detect 0x000001 as H264 byte-stream start code in codec_data
+ This works around some AVI files storing byte-stream data in the
+ codec_data. The previous workaround was only checking for
+ 0x00000001 (4 bytes) instead of 0x000001 (3 bytes).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1072>
+
+2021-08-31 11:05:16 +0100 Philippe Normand <philn@igalia.com>
+
+ * ext/qt/qtitem.cc:
+ qt: Fix build for Qt 5.9
+ The QQuickItem::size() method was introduced in 5.10, so use direct width() and
+ height() access instead.
+ Fixes #908
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1069>
+
+2021-08-19 10:32:27 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/elements/rtphdrextrfc6464.c:
+ rtphdrext-rfc6464: Add test for inserting in payloader using the API
+ This makes it clearer how to use the plugin in an API driven application.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
+
+2021-08-18 19:36:07 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtphdrext-rfc6464.c:
+ rtphdrext-rfc6464: Put max level if the audio is beyond it
+ Otherwise, it just fails to add the extension, which makes no
+ sense. And our level element produces levels higher than 127 in some
+ cases.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
+
+2021-08-18 19:35:36 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtphdrext-rfc6464.c:
+ rtphdrext-rfc6464: Add example pipeline
+ This makes it a bit easier to understand how to use it in an application.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
+
+2021-08-18 19:07:18 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/elements/rtphdrextrfc6464.c:
+ rtphdrext-rfc6464: Add test for inserting it based on caps
+ Tests adding the extension based on the caps.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
+
+2021-08-27 14:32:45 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Force stream-start push when re-using EOS'd streams
+ When re-using streams, we *do* need to push a `stream-start` event downstream if
+ we previously were EOS'd. Failure to do that would never remove the EOS status
+ on all downstream elements and cause weird issues.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1067>
+
+2021-08-27 02:05:45 -0400 Brad Smith <brad@comstyle.com>
+
+ * gst/deinterlace/meson.build:
+ deinterlace: Use proper ASM output format for *BSD OS
+ FreeBSD/NetBSD/OpenBSD amd64 use the ELF binary format.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1066>
+
+2021-08-27 13:30:57 +1000 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/qtitem.cc:
+ qmlgl: don't critical on input events before input format has been set
+ Accessing the unset GstVideoInfo would result in criticals
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1065>
+
+2021-08-25 11:53:58 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ docs: Add `Since` marker to "twcc-feedback-interval" property
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-08-25 10:33:24 +0200 Havard Graff <havard@pexip.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: update with "twcc-feedback-interval"
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-04-13 16:19:22 +0200 Tulio Beloqui <tulio.beloqui@pexip.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpstats.h:
+ * gst/rtpmanager/rtptwcc.c:
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: changes to use rtp buffer arrival time and current time.
+ For TWCC we are more interested to track the arrival time (receive side)
+ and the current time (sender side) of the buffers rather than the
+ running time.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-03-26 11:57:42 +0100 Knut Inge Hvidsten <knut.hvidsten@pexip.com>
+
+ * gst/rtpmanager/rtpstats.c:
+ * gst/rtpmanager/rtptwcc.c:
+ * gst/rtpmanager/rtptwcc.h:
+ rtptwcc: add payloadtype to RTPTWCCPacket
+ The consumer of the stats can then separate between different media-types,
+ and do individual stats for each of them.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-03-19 18:19:43 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ rtptwcc: make enabling TWCC sticky
+ Meaning that if a caps comes along that does NOT have TWCC in it,
+ this does not turn of TWCC for the rest, as this is in fact
+ completely allowed. (To have some payload-types not containing TWCC
+ seqnums).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-02-23 09:44:05 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtptwcc.c:
+ * gst/rtpmanager/rtptwcc.h:
+ rtptwcc: move TWCC-logic over to the TWCC-manager
+ Prevent cluttering up the rtpsession, and keeping things localized.
+ Also write TWCC-seqnums for *all* streams in the session if configured by
+ caps.
+ A while back WebRTC was not doing TWCC for audio, basically breaking the
+ whole idea of a "transport-wide seqnuencenumber" applying for all bundled
+ streams. However, they have since fixed this, and now it no longers
+ makes sense to be able to single out certain payloadtypes for
+ use with TWCC, rather just including them all.
+ This also makes using RTX, RED, FEC etc much simpler, as it will apply
+ to them all as they enter the rtpsession.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-02-23 09:50:04 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ rtptwcc: fix warning
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-02-11 15:17:16 +0100 Tulio Beloqui <tulio.beloqui@pexip.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: fixes and optimizations around run-length chunks
+ Co-authored-by: Havard Graff <havard.graff@gmail.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2020-12-18 14:01:23 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: fix seqnum-wrap
+ Using the proper API to do this is obviously an improvement, and
+ adding a test for the case of a packet-loss when the seqnum wrap
+ is also a good idea.
+ Co-authored-by: Tulio Beloqui <tulio.beloqui@pexip.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2020-12-18 13:06:35 +0100 Tulio Beloqui <tulio.beloqui@pexip.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: fixed feedback packet count overflow that allowed late packets to be processed
+ Co-authored-by: Havard Graff <havard.graff@gmail.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2020-12-16 16:31:18 +0100 Tulio Beloqui <tulio.beloqui@pexip.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: fixed parsing of old sequence number
+ Co-authored-by: Havard Graff <havard.graff@gmail.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2020-12-16 16:16:09 +0100 Tulio Beloqui <tulio.beloqui@pexip.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: fixed guint8 overflow of feedback packet count
+ Co-authored-by: Havard Graff <havard.graff@gmail.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2020-11-19 23:50:23 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtptwcc.c:
+ * gst/rtpmanager/rtptwcc.h:
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: add feedback-interval
+ To allow RTCP TWCC reports to be scheduled on a timer instead of per
+ marker-bit.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-08-20 11:54:01 +0200 Havard Graff <havard@pexip.com>
+
+ * gst/rtpmanager/rtptwcc.c:
+ * gst/rtpmanager/rtptwcc.h:
+ rtptwcc: remove _set_send_packet_ts
+ Not in use.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2020-11-17 00:45:02 +0100 Havard Graff <havard@pexip.com>
+
+ * tests/check/elements/rtpsession.c:
+ rtptwcc: make twcc-tests more deterministic
+ They were a bit racy.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/927>
+
+2021-08-24 13:28:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add depth for ProRes 4:4:4:4 variants if available
+ Might be 24bpp in case an alpha channel is coded but
+ the image is always opaque.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1061>
+
+2021-08-22 23:16:26 +0000 Ruslan Khamidullin <ruslank@borisfx.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: for Apple ProRes, allow overriding pixel bit depth for 4:4:4:4 variants
+ e.g. when exporting an opaque image, yet with alpha channel.
+ Apple ProRes certification requires that, when a ProRes-writing
+ application *knows* that the entire frame is opaque, the application
+ writes only RGB without alpha even when the clip is RGBA. For that,
+ this tiny change allows the app to override pixel depth when writing ProRes.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1061>
+
+2019-05-22 11:16:56 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * ext/vpx/gstvpxdec.c:
+ * ext/vpx/gstvpxdec.h:
+ vpxdec: Fix direct rendering, avoid holding write access
+ When a buffer is pushed downstream, we should try not to hold the
+ buffer mapped with write access. Doing so would often lead to
+ an unneccesary memcpy later.
+ For instance, gst_buffer_make_writable() in
+ gst_video_decoder_finish_frame() will cause a memcpy because of
+ _memory_get_exclusive_reference().
+ We know that we can perform a two-step remap when using system
+ memory, as this will not cause the location of the memory to
+ change.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/812>
+
+2021-08-19 16:26:17 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4/mux: add a function for seeking to a specific output byte position
+ We do it enough times that this makes sense. Also add a debug log line
+ for the seek position requested.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>
+
+2021-08-19 16:02:47 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4/mux: don't overwrite with a bigger moov when fragmenting
+ When outputting fragmented mp4, with a seekable downstream, we rewrite
+ the moov to maybe add a duration to the mvex. If we start by not
+ writing the initial moov->mvex->mhed duration and then overwrite with a
+ moov containing mhed atom, the moov's will have different sizes and
+ could overwrite subsequent data and result in an unplayable file.
+ e.g. The initial moov would be of size 842 and the final moov would have
+ a size of 862.
+ Fix by always pushing out the mhed duration in the moov when
+ fragmenting.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/898
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>
+
+2021-01-15 20:53:27 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: actually make streamable fallback work
+ We weren't setting the fragment_mode field anymore now that the
+ implementation doesn't change based on the value of the streamable
+ property. This lead to invalid files.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>
+
+2021-01-15 20:54:56 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4: fix trun data offset handling
+ The trun offset was missing a calculation for one of the box type
+ headers.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/866
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>
+
+2020-10-15 00:28:36 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/gstqtmux.c:
+ isomp4/mux: fixes for fragmented mp4 output
+ Various buffer offset calculations were not quite correct in all cases.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/866
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1060>
+
+2021-08-10 00:53:57 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: support H264 avc3 / H265 hev1
+ The matroska codec specs is unfortunately vague on the subject,
+ stating for H264:
+ AVC/H.264 stored as described in [@!ISO.14496-15]
+ and for H265:
+ HEVC/H.265 stored as described in [@!ISO.14496-15]
+ This spec however specifies multiple stream formats, our
+ implementation has opted for interpreting this as avc1 / hvc1,
+ both of which disallow in-band SPS.
+ Most decoders however will support in-band SPS / PPS, and
+ this commit gives the option to explicitly mux in avc3 / hev1,
+ which allows changing stream parameters on the fly, that is
+ useful for smart encoding for example.
+ When either of these stream formats are picked as the input,
+ changes in codec_data / tier / level / profile do not cause
+ renegotiation failure, a warning is logged however as it isn't
+ clear how compliant such a stream is.
+ The stream-format field is correctly ordered in the template
+ caps to avoid selecting potentially non-compliant options on
+ automatic negotiation.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
+
+2021-08-10 00:51:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4/qtmux: allow renegotiating when tier / level / profile change
+ Those are carried either in codec_data or in-band SPS (for avc3),
+ and it is OK for those to change, though decoders obviously need
+ to support it.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
+
+2021-08-06 23:36:48 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ isomp4/qtmux: accept video/x-h264, stream-format=avc3
+ The main difference between avc1 and avc3 is that avc3 is allowed
+ to contain in-band SPS / PPS. In practice decoders will always use
+ in-band parameter sets anyway, but it is cleaner to explicitly
+ advertise it.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
+
+2021-08-06 22:59:23 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ isomp4/qtmux: make sure to switch to next chunk on new caps
+ For example, with single video sink pad, and new codec_data is
+ received, current_chunk_offset must be reset to -1 for the
+ aggregate loop to open a new chunk.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
+
+2021-08-06 22:55:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4/atoms: fix multiple stsd entries
+ stsd entries are serialized in reverse order (starting from
+ g_list_last()), and must be prepended to the entry list for their
+ index to be correct when referenced from stsc entries.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1047>
+
+2021-08-12 11:03:58 -0400 Arun Raghavan <arun@asymptotic.io>
+
+ * docs/gst_plugins_cache.json:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroska-mux: Add a timestamp-offset property
+ Adds a user-controllable timestamp offset to clusters and blocks. This
+ should be useful if we want to have timestamps that have significance
+ outside of the current file (for example, we might set the offset to the
+ wallclock when the file is being created, or some other common base, if
+ we want to correlate streams across multiple files).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1051>
+
+2021-07-15 12:02:40 +0200 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska: demux: update stream_start_time
+ The stream_start_time can be less than the first detected.
+ In case of B-Frame based media, the first frame PTS might be
+ greater than the next one.
+ Need to keep the segment.start if a seek has been performed.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>
+
+2021-08-17 16:08:33 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/matroska/matroska-demux.c:
+ mastrokademux: Remove redundant assignment
+ The segment.position is unconditionnaly set few lines below.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>
+
+2021-08-17 16:49:47 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ * tests/icles/videocrop-test.c:
+ videocrop: Fix icles tests.
+ Internally videcrop can call gst_video_crop_set_info() with NULL as in
+ caps. Then critical messages are raised when the in caps are
+ processed.
+ To fix this the in caps are checked, and if they are present, its
+ capsfeature is extracted, otherwise, the previous raw caps detection
+ remains as before.
+ Also the videocrop-test removes the format field in the structure
+ because now its always passed.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1056>
+
+2021-01-08 17:34:02 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpelements.h:
+ * gst/rtp/gstrtphdrext-colorspace.c:
+ * gst/rtp/gstrtphdrext-colorspace.h:
+ * gst/rtp/meson.build:
+ * tests/check/elements/rtphdrext-colorspace.c:
+ * tests/check/meson.build:
+ rtp: Color Space header extension
+ Implements WebRTC header extension defined in
+ http://www.webrtc.org/experiments/rtp-hdrext/color-space.
+ It uses RTP header to communicate color space information and optionally
+ also metadata that is needed in order to properly render a high dynamic
+ range (HDR) video stream.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/853>
+
+2021-08-09 10:46:30 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Add protection when set decoder capture fps accroding to output fps
+ Some v4l2 drivers don't have the capacity to change framerate. There is
+ chance to make decoder capture fps to be 0/0 if numerator and denominator
+ returned by G_PARM ioctl are both 0. It causes critical warning
+ "passed '0' as denominator for `GstFraction'".
+ In order to fix this, add protection when set decoder capture fps according
+ to output fps.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1048>
+
+2021-06-01 15:33:01 +0200 Per Förlin <perfn@axis.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Add support to ignore x-server HEADER reply
+ When connecting to an RTSP server in tunnled mode (HTTP) the server
+ usually replies with a x-server header. This contains the address
+ of the intended streaming server. However some servers return an
+ "invalid" address. Here follows two examples when it might happen.
+ 1. A server use Apache combined with a separate RTSP process to handle
+ Https request on port 443. In this case Apache handle TLS and
+ connects to the local RTSP server, which results in a local
+ address 127.0.0.1 or ::1 in the x-server reply. This address is
+ returned to the actual RTSP client in the x-server header.
+ The client will receive this address and try to connect to it
+ and fail.
+ 2. The client use a ipv6 link local address with a specified scope id
+ fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
+ The RTSP server receives the connection and returns the address
+ in the x-server header. The client will receive this address and
+ try to connect to it "as is" without the scope id and fail.
+ In the case of streaming data from RTSP servers like 1. and 2. it's
+ useful to have the option to simply ignore the x-server header reply
+ and continue using the original address.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1007>
+
+2021-08-04 12:33:06 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * ext/qt/qtitem.cc:
+ gstqmlgl: fix indent
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-07-30 16:52:23 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * ext/qt/gstqsgtexture.cc:
+ * ext/qt/gstqsgtexture.h:
+ gstqmlgl: wrap raw GstGLContext into GWeakRef
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-07-30 16:32:13 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * tests/examples/qt/meson.build:
+ * tests/examples/qt/qmlsink-multisink/main.cpp:
+ * tests/examples/qt/qmlsink-multisink/main.qml:
+ * tests/examples/qt/qmlsink-multisink/meson.build:
+ * tests/examples/qt/qmlsink-multisink/qmlsink-multi.qrc:
+ * tests/examples/qt/qmlsink-multisink/videoitem/VideoItem.qml:
+ * tests/examples/qt/qmlsink-multisink/videoitem/videoitem.cpp:
+ * tests/examples/qt/qmlsink-multisink/videoitem/videoitem.h:
+ gstqmlgl: add multisink test application
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-07-30 17:21:46 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * ext/qt/qtitem.cc:
+ * ext/qt/qtitem.h:
+ gstqmlgl: refactoring: rename ambiguous variables, clean up unused and duplicated ones
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-07-30 17:20:59 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * ext/qt/gstqtglutility.cc:
+ gstqmlgl: rework WGL-specific context init code
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-07-30 17:20:49 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * ext/qt/gstqtglutility.cc:
+ gstqmlgl: retrieve correct device bound to current GL context (+ minor code cleanup)
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-07-30 17:20:25 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * ext/qt/gstqsgtexture.cc:
+ gstqmlgl: correct validation for Qt GL context
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-07-30 17:20:07 +0700 Dmitry Shusharin <pmdvsh@gmail.com>
+
+ * ext/qt/gstqtglutility.h:
+ * ext/qt/qtitem.cc:
+ * ext/qt/qtitem.h:
+ * ext/qt/qtwindow.cc:
+ * ext/qt/qtwindow.h:
+ gstqmlgl: create helper QRunnable-based class for render jobs
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
+
+2021-08-06 16:25:02 +0200 Tulio Beloqui <tulio@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fixed stall on gap when using rtx
+ Co-authored-by: Håvard Graff <havard@pexip.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1055>
+
+2021-08-13 19:32:53 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/flv/gstindex.c:
+ flv: use g_memdup2() as g_memdup() is deprecated
+ g_memdup() is deprecated since GLib 2.68 and we want to avoid
+ deprecation warnings with recent versions of GLib.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1052>
+
+2021-08-15 12:26:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Always use the content decoder but set `Accept-Encoding: identity` if no compression should be used
+ Some servers respond with gzip-encoded responses regardless of whether
+ the request allowed it to be used in the response. By always having the
+ content decoder enabled, these invalid responses can be decoded
+ correctly while for well-behaving servers the `compress` property
+ selects between allowing compressed responses or not.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/833
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1053>
+
+2021-08-12 22:57:01 +1000 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/gstqtglutility.cc:
+ qt: always update the sink_retrieved flag when the sink retrieves
+ Fixes a case where adding a qmlgloverlay element after an existing
+ qmlglsink elements was already in the pipeline would create an entirely
+ separate GstGLDisplay pointing to the same underlying display resource.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1050>
+
+2021-08-11 14:52:52 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstaspectratiocrop.h:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videocrop/gstvideocrop.h:
+ * gst/videocrop/gstvideocropelement.c:
+ * gst/videocrop/gstvideocropelements.h:
+ * gst/videocrop/gstvideocropplugin.c:
+ * gst/videocrop/meson.build:
+ videocrop: Resurrect logging category.
+ Fix for a regression from commit 8f1384c9. That commit moved the debug
+ category definition, as static, into a gstvideocropelement.c, but that
+ category was used as default, in gstvideocrop.c, so it was never used
+ at logging, so the debug selector never showed the logs for
+ videocrop.
+ This patch move back the category definition into gstvideocrop.c and
+ leaving the function videocrop_element_init() as a noop.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1049>
+
+2021-07-31 23:14:34 +0900 Seungha Yang <seungha@centricular.com>
+
+ * ext/jpeg/meson.build:
+ jpeg: Add support for meson fallback
+ Allow building jpeg plugin by using meson fallback
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1041>
+
+2021-07-27 10:43:21 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Keep decoder capture fps same as output fps if it's not set
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1035>
+
+2021-07-27 18:33:18 +0900 Seungha Yang <seungha@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/jack/gstjack.c:
+ * ext/jack/gstjack.h:
+ * ext/jack/gstjackaudioclient.c:
+ * ext/jack/gstjackaudioclient.h:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosink.h:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/jack/gstjackaudiosrc.h:
+ jack: Add port-names property to select ports explicitly
+ By this new property, user can select physical port to connect,
+ and element will pick requested port instead of random ones.
+ User should provide full port name including "client_name:" prefix.
+ An example is
+ jackaudiosrc port-names="system:capture_1,system:capture_3" ! ...
+ jackaudiosink port-names="system:playback_2"
+ In addition to "port-names" property, a new connect type "explicit"
+ is added so that element can post error message if requested
+ "port-names" contains invalid port(s).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1037>
+
+2021-07-23 11:04:00 +0200 Kai Uwe Broulik <kai_uwe.broulik@mbition.io>
+
+ * ext/qt/gstqsgtexture.cc:
+ * ext/qt/gstqtsink.cc:
+ qt: Support RGB format
+ In GstQSGTexture::hasAlphaChannel return value based on
+ whether the video format has alpha channel.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1040>
+
+2021-07-26 20:14:32 +0900 Seungha Yang <seungha@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosink.h:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/jack/gstjackaudiosrc.h:
+ jack: Add low-latency property for automatic latency-optimized setting
+ Similar to wasapi/wasapi2 plugins on Windows, adding low-latency
+ option so that jack element can optimize GstAudioRingBufferSpec
+ setting for low latency.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1034>
+
+2021-07-26 19:55:25 +0900 Seungha Yang <seungha@centricular.com>
+
+ * ext/jack/gstjackaudioclient.c:
+ * ext/jack/gstjackaudioclient.h:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/jack/gstjackringbuffer.h:
+ jack: Remove trailing whitespace
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1034>
+
+2021-07-27 17:58:15 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/videocrop/gstvideocrop-private.h:
+ videocrop: Resurrect any caps feature negotiation.
+ Commit e31cbce4 brought a regression to negotiate featured caps. But
+ only by removing the entry in the caps template. This commit brings it
+ back.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1039>
+
+2021-07-26 18:43:04 +0900 Seungha Yang <seungha@centricular.com>
+
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosrc.c:
+ jack: Fix assertion fail when device supports only mono channel
+ MAX should be larger than MIN for GST_TYPE_INT_RANGE.
+ GStreamer-CRITICAL **: 18:26:27.912:
+ gst_value_collect_int_range: assertion 'collect_values[0].v_int < collect_values[1].v_int' failed
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1033>
+
+2021-07-21 20:14:46 +1000 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/gstqtsrc.cc:
+ * ext/qt/qtwindow.cc:
+ * ext/qt/qtwindow.h:
+ qmlglsrc: fix operation without any qmlglsink
+ E.g. a pipeline like qmlglsrc ! gldownload ! ... would currently fail to
+ run because the OpenGL context are not created in the correct order.
+ The QtWindow also needs to know the OpenGL context used by downstream
+ elements in order to set optimize for the correct GstGLSyncMeta for
+ synchonisation purposes.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1036>
+
+2021-07-26 17:55:24 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix some reference leaks in error cases.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>
+
+2021-07-08 00:12:52 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * tests/check/elements/splitmuxsink.c:
+ splitmuxsink: Prevent hang going back to NULL after failures
+ Prevent a condition where splitmuxsink won't go back to NULL state
+ after a child element fails to change state by making sure that
+ a READY->READY state change doesn't fail, and by returning
+ GST_FLOW_ERROR or GST_FLOW_FLUSHING upstream to shut down streaming
+ as quickly as possible.
+ This can happen after (for example) setting an invalid filename
+ on the sink element. In that case, the READY->PAUSED transition
+ fails, but with internal elements still in the NULL state. Trying
+ to set splitmuxsink back to NULL then ends up trying to bring
+ those NULL elements up to READY with a READY->READY transition,
+ (which fails, prevent splitmuxsink from getting to NULL)
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1023>
+
+2021-07-13 01:27:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: reduce noise when gst_pad_set_caps fails
+ It may be that downstream is simply flushing, in which case logging
+ an error is misleading.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1029>
+
+2021-07-08 02:22:20 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: always use factory property when set
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1024>
+
+2021-06-21 13:47:50 +0200 Yacine Bandou <yacine.bandou@softathome.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: No need for new "application/x-cbcs" caps
+ Instead of using the new "application/x-cbcs" caps, we are just adding
+ a new structure field "ciphe-mode", to indicate which encryption scheme
+ is used: "cenc", "cbcs", "cbc1" or "cens".
+ Similarly for the protection metadata, we add the "cipher-mode" field
+ to specify the encryption mode with which the buffers are encrypted.
+ "cenc": AES-CTR (no pattern)
+ "cbc1": AES-CBC (no pattern)
+ "cens": AES-CTR (pattern specified)
+ "cbcs": AES-CBC (pattern specified, using a constant IV)
+ Currently only "cenc" and "cbcs" are supported.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1013>
+
+2021-07-05 16:12:57 +0100 Philippe Normand <philn@igalia.com>
+
+ * ext/qt/qtitem.h:
+ qt: Fix clang build
+ The updatePaintNode method is part of the QQuickItem class interface, so needs
+ to be flagged as overriding the default implementation.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/567>
+
+2020-04-15 10:38:04 +0100 Philippe Normand <philn@igalia.com>
+
+ * ext/qt/gstqtsink.cc:
+ * ext/qt/qtitem.cc:
+ * ext/qt/qtitem.h:
+ qt: Add navigation events support
+ Currently handles only mouse events.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/567>
+
+2020-04-15 10:33:22 +0100 Philippe Normand <philn@igalia.com>
+
+ * ext/gtk/gtkgstbasewidget.c:
+ gtk: Scroll events dispatch support
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/567>
+
+2021-07-02 01:41:05 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Always write a tags element into seekhead
+ If there are only stream tags, we still want to write the
+ tags entry into the seekhead, so that tags can be found
+ quickly in the player.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/905
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1020>
+
+2021-06-30 23:52:26 +0900 Seungha Yang <seungha@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't need to update track per GstCaps if it's not changed
+ Skip GstQTMuxPad::set_caps() call for duplicated caps.
+ All the processing done in set_caps() method for duplicated caps
+ are redundant.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1019>
+
+2021-07-01 13:18:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ rtpssrcdemux: Remove pads and reset the element also in READY->NULL
+ Mostly for completeness.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>
+
+2021-07-01 13:18:09 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpptdemux.c:
+ rtpptdemux: Remove pads also in PAUSED->READY
+ They're based on per-stream information and that should be reset
+ whenever going to READY state.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1018>
+
+2021-02-16 16:39:34 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: update plugins cache for vp9enc
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>
+
+2021-04-09 19:22:29 +0200 Jakub Adam <jakub.adam@collabora.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ * ext/vpx/gstvpxenums.h:
+ * ext/vpx/meson.build:
+ vpx: add enum for adaptive quantization modes
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>
+
+2021-02-16 13:28:00 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp9enc: expose frame-parallel-decoding property
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>
+
+2021-02-16 12:57:55 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp9enc: expose aq-mode property
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/874>
+
+2021-06-26 20:00:03 +0900 Seungha Yang <seungha@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: Fix broken SO_SNDBUF get/set on Windows
+ SO_SNDBUF has been undefined on Windows because of missing WinSock2.h
+ include. And don't use native socket functions (e.g., setsockopt())
+ if code is expected to be built on Windows. We don't link ws2_32.lib
+ for this plugin.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1016>
+
+2021-06-24 14:57:14 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtphdrext-twcc.c:
+ rtpmanager: Access GstRTPHdrExt fields through accessor
+ This way, the implementation can be private.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1017>
+
+2021-06-22 17:19:19 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Refuse seeks in BYTES format
+ If downstream tries to seek in BYTES format, don't pass that through
+ to upstream. The byte positions downstream requests won't make any
+ sense in the muxed stream. There might be other formats we want to
+ pass through to upstream, but BYTES is not one of them. If we get a
+ seeking query about BYTES format, refuse that too.
+ This fixes a situation where we're playing a fragmented mp4 over http
+ and qtdemux refuses the initial seek (in TIME format), but then
+ h264parse/baseparse send a seek in BYTES format and everything falls
+ apart.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1014>
+
+2021-06-16 16:30:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtp/gstrtph265depay.c:
+ rtph265depay: update codec_data in caps regardless of format
+ Updating of codec_data in the caps is important to propagate changes
+ in sps/pps/vps via NALs. Without this, downstream does not renegotiate
+ when upstream changes resolution.
+ The comment referring to rtph264pay is from 2015 and is out of date.
+ rtph264pay stopped doing that in 2017 with commit
+ dabeed52a995d27e16eba9e4617e61eb0bcd44c4
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1011>
+
+2021-06-04 13:56:05 +0300 Jordan Petridis <jordan@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ doc: update gst_plugins_cache.json
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1006>
+
+2021-06-03 20:33:45 +1000 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/qtitem.cc:
+ qtitem: don't potentially leak a large number of buffers
+ The only other place where these queued buffers are removed, is in
+ setCaps() but that is not called at all on shutdown so this list of
+ buffers could not be removed.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1004>
+
+2021-05-28 09:54:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/jpeg/gstjpegenc.c:
+ jpegenc: Remove arbitrary encoding size limitation
+ The encoder is happy to encode with sizes less then 16x16, so remove this
+ arbitrary limitation. This also fixes the fact the sink and src template caps
+ disagree.
+ Fixes #888
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/998>
+
+2021-05-23 15:42:38 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: use g_memdup2() as g_memdup() is deprecated
+ - atom nodes/bytereader sizes are already checked
+ - palettes: are fixed/known size
+ g_memdup() is deprecated since GLib 2.68 and we want to avoid
+ deprecation warnings with recent versions of GLib.
+ Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>
+
+2021-05-23 01:28:11 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/ebml-read.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: use g_memdup2() as g_memdup() is deprecated
+ - ebml-read: add some sanity checks when going from 64-bit
+ to 32-bit length
+ - matroska-ids: codec_data_size has been checked via
+ gst_ebml_read_binary(), is existing allocation.
+ - matroska-demux: alloc size is from existing allocations
+ g_memdup() is deprecated since GLib 2.68 and we want to avoid
+ deprecation warnings with recent versions of GLib.
+ Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>
+
+2021-05-22 19:39:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/libpng/gstpngdec.c:
+ * ext/vpx/gstvpxenc.c:
+ * ext/wavpack/gstwavpackenc.c:
+ * meson.build:
+ * tests/check/elements/rtp-payloading.c:
+ * tests/check/elements/rtph264.c:
+ * tests/check/elements/rtph265.c:
+ * tests/check/elements/rtpopus.c:
+ * tests/check/elements/rtpvp8.c:
+ * tests/check/elements/rtpvp9.c:
+ * tests/icles/gdkpixbufoverlay-test.c:
+ Use g_memdup2() where available and add fallback for older GLib versions
+ - png: alloc size variable is a png type that's always 32-bit
+ - vpx: alloc size based on existing allocation
+ - wavpack: alloc size based on existing allocation
+ - icles: gdkpixbufoverlay: trusted and hard-coded input data
+ - rtp tests: rtp-payloading, vp8, vp9, h264, h265: trusted and/or static input data
+ g_memdup() is deprecated since GLib 2.68 and we want to avoid
+ deprecation warnings with recent versions of GLib.
+ Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..)
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>
+
+2021-06-01 15:28:36 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
+=== release 1.19.1 ===
+
+2021-06-01 00:11:44 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * README:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-plugins-good.doap:
+ * meson.build:
+ Release 1.19.1
+
+2021-05-29 12:54:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpjpegpay.c:
+ rtpjpegpay: fix image corruption when compiled with MSVC on Windows
+ On Windows with MSVC, jpeg_header_size would end up 2 bytes larger
+ than it should be. This then leads to the first 2 bytes of the
+ actual jpeg image data to be dropped, because we think those
+ belong to the header, which results in an undecodable image when
+ reconstructed in the depayloader.
+ What happens is that when the compiler evaluates
+ jpeg_header_size = mem.offset + read_u16_and_inc_offset_by_2(&mem);
+ it actually uses the mem.offset value after it has been increased
+ by the function call on the right hand size of the equation.
+ From section 6.5 of the C99 spec:
+ 3. The grouping of operators and operands is indicated by the syntax [74].
+ Except as specified later (for the function-call (), &&, ||, ?:, and
+ comma operators), the order of evaluation of subexpressions and the
+ order in which side effects take place are both unspecified.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/889
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/999>
+
+2021-05-25 16:19:20 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2videoenc.c:
+ v4l2videoenc: Set default latency if the frame duration is invalid
+ If the duration of the v4l2object is invalid, use default 25fps instead.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/996>
+
+2021-05-26 00:23:56 +0900 Seungha Yang <seungha@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Drop "field-order" field while transforming caps
+ Like other basetransform subclasses are doing, drop field
+ which can be converted by deinterlace.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>
+
+2021-05-25 20:10:34 +0900 Seungha Yang <seungha@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Drop field-order field if outputting progressive
+ Progressive with field-order doesn't make sense
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>
+
+2021-05-21 14:19:29 +0200 Havard Graff <havard@pexip.com>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * tests/check/elements/rtpssrcdemux.c:
+ rtpssrcdemux: fix "data flow before segment event" crash
+ This crash could happen at any time a RTP and RTCP buffer arrived
+ simultaneously in ssrcdemux.
+ The problem was that sticky-event arriving while the rtp and rtcp pads
+ were being set up could arrive just too late to be included in the initial
+ forwarding.
+ The fix checks if the stickies have been sent on the srcpad about to be
+ pushed on, and if not sends them. It also blocks any stickes from
+ being forwarded *prior* to this happening, to avoid them arriving on
+ the srcpad multiple times.
+ Since the test loops 1000 times, this will make running under valgrind
+ take forever, so use the RUNNING_ON_VALGRIND variable to detect we
+ are running under valgrind, and reduce the loop-count to 2 in that case.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
+
+2021-05-21 18:45:17 +0200 Havard Graff <havard@pexip.com>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ rtpssrcdemux: refactor destruction of GstRtpSsrcDemuxPads
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
+
+2021-05-21 18:30:28 +0200 Havard Graff <havard@pexip.com>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/rtpmanager/gstrtpssrcdemux.h:
+ rtpssrcdemux: make naming consistent
+ Use plural for GstRtpSsrcDemuxPads, since it contains two pads, and
+ use the variable-name 'dpads' everywhere.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
+
+2021-05-23 15:14:11 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: use g_strndup() for copying text data
+ So we don't rely on NUL terminators inside the data.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
+
+2021-05-23 13:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: clean up adtl/note/labl chunk parsing
+ We were passing the size of the adtl chunk to the note/labl
+ sub-chunk parsing function, which means we may memdup lots of
+ data after the chunk string's NUL terminator that doesn't
+ really belong to it.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
+
+2021-05-23 13:24:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: guard against overflow when comparing chunk sizes
+ Could be rewritten as lsize > (size - 8) a well, but the
+ extra check seems clearer. Doesn't look like it was problematic,
+ lsize wasn't actually used when parsing the sub-chunks.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
+
+2021-05-21 13:31:12 -0300 Daniel Almeida <daniel.almeida@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ doc: update gst_plugins_cache.json
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/991>
+
+2021-05-05 13:20:04 +0200 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix decoder glitches with H264 content
+ To avoid decoder starvation causing glitches on screen,
+ the demuxer shall clip only when the buffer is a key frame
+ and the lace time is greater than the stop time.
+ Fixes gst-editing-services#128
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/973>
+
+2021-05-11 20:41:38 +1000 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/gstqtoverlay.cc:
+ qml: don't use buffers that have invalid contents
+ If the GL context is not shareable, ignore it.
+ A future change may also not output the relevant output either.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>
+
+2021-05-11 20:38:52 +1000 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/gstqsgtexture.cc:
+ qml: also use the dummy texture when no buffer has been set
+ Fixes corrupted texture output when changing OpenGL display/contexts.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>
+
+2021-05-11 17:20:00 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ doc: Update cache for RGBP format addition
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2021-04-23 14:37:46 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: Advertise codec-alpha in caps
+ This will be used to select the appropriate decoders. We also only attach the
+ GstVideoCodecAlphaMeta if the AlphaMode element is set, this is to stay on the
+ safe side and mimic what browsers (verified in Firefox and Chromium code) do.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2021-03-22 16:58:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Store alpha stream in VideoCodecAlphaMeta
+ This generalize the feature over using mini object quark data. If
+ that feature was Matroska specifc, using the new CustomMeta would have
+ been enough and arguably cleaner then QData, though it seems that
+ similar technique is use with AV1 Image Format (AVIF).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2016-12-03 14:27:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: extract VP8 alpha from BlockAdditionals
+ And put it on buffers as qdata (which is easier in this
+ case than a private custom meta because it can be picked
+ up easily in other modules).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2021-05-03 17:39:05 +1000 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/gstqtglutility.cc:
+ * ext/qt/gstqtglutility.h:
+ * ext/qt/gstqtoverlay.cc:
+ * ext/qt/qtitem.cc:
+ * ext/qt/qtwindow.cc:
+ qt: return a different GstGLDisplay object when the first sink requests
+ This allows the 'replace-gstreamer-opengl-context' context machinery to
+ correctly replace the OpenGL context used by the pipeline when the first
+ qmlglsink is added to the pipeline.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/982>
+
+2021-05-07 11:16:47 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Plug leaks of saddr in error cases
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>
+
+2021-05-07 11:16:21 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Whitespace
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>
+
+2021-05-07 00:43:44 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Plug a method subobject leak
+ Changing the method would leak the previous method.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/976>
+
+2021-05-06 15:04:42 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvp9enc.c:
+ vp9enc: Add color range support
+ When setting the colorspace, we now clear the range to reduced range,
+ the default, and then we also set the range so the VP9 encoder encodes
+ the right information in the bitstream.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>
+
+2021-05-06 14:51:31 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxenc.c:
+ vp9enc: Move colorspace configuration in VP9 enc
+ This is not supported by VP8 and was causing a warning.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>
+
+2021-05-06 14:48:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxenc.c:
+ * ext/vpx/gstvpxenc.h:
+ vpxdenc: Add a GstVideoCodecState to configure_encoder virtual
+ This will be needed to configure the VP9 specific colorimetry, which is
+ currently configured for VP8 casing warning.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>
+
+2021-05-05 16:48:10 +0200 Bastien Nocera <hadess@hadess.net>
+
+ * ext/gtk/gtkgstbasewidget.c:
+ gtk: Remove coordinates double-translation
+ Remove our own translation in the mouse event capture code, as that
+ translation will be done through the navigation interface.
+ Tested by resizing the window created by:
+ gst-launch-1.0 -v videotestsrc ! navigationtest ! glupload ! glcolorconvert ! tee name=t ! gtkglsink
+ and checking that the cursor follows the mouse as expected.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>
+
+2021-05-05 14:28:15 +0200 Bastien Nocera <hadess@hadess.net>
+
+ * ext/gtk/gstgtkbasesink.c:
+ gtk: Translate navigation events coordinates
+ If the application passed down some pointer coordinates, translate those
+ from display coordinates to stream coordinates, so things work as
+ expected even if the video is resized.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>
+
+2021-05-05 14:24:31 +0200 Bastien Nocera <hadess@hadess.net>
+
+ * ext/gtk/gtkgstbasewidget.c:
+ * ext/gtk/gtkgstbasewidget.h:
+ gtk: Export _display_size_to_stream_size()
+ Export _display_size_to_stream_size() so that GstNavigation implementors
+ can translate from display coordinates to stream coordinates before
+ pushing the events upstream to the DVD source, for example.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>
+
+2018-02-26 17:26:41 +0100 David Fernandez <d.fernandezlop@gmail.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Change accepted caps width and height from [16, MAX] to [1, MAX]
+ There are cases where the video size might be less than 16x16.
+ This change allows the Matroska muxer to accept this cases.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/539>
+
+2021-04-20 22:08:23 +0200 François Laignel <fengalin@free.fr>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * tests/check/elements/avimux.c:
+ * tests/check/elements/flvmux.c:
+ * tests/check/elements/interleave.c:
+ * tests/check/elements/qtmux.c:
+ * tests/check/elements/rtpbin.c:
+ * tests/check/elements/rtpcollision.c:
+ * tests/check/elements/rtpmux.c:
+ * tests/check/elements/splitmuxsink.c:
+ * tests/check/elements/videomixer.c:
+ * tests/examples/rtp/client-PCMA.c:
+ * tests/examples/rtp/server-alsasrc-PCMA.c:
+ Use gst_element_request_pad_simple
+ Instead of the deprecated gst_element_get_request_pad.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/958>
+
+2021-04-30 08:12:47 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ qtmux: Make sure to write 64-bit STCO table when needed.
+ qtmux attempts to choose between writing a 32-bit stco chunk offset table
+ when it can, but switch to a 64-bit co64 table when file offsets go over
+ 4GB.
+ This patch fixes a problem where the atom handling code was checking
+ mdat-relative offsets instead of the final file offset (computed by
+ adding the mdat position plus the mdat-relative offset) - leading to
+ problems where files with a size between 4GB and 4GB+offset-of-the-mdat
+ would write incorrect STCO tables with some samples having truncated
+ 32-bit offsets.
+ Smaller files write STCO correctly, larger files would switch to
+ co64 and also output correctly.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/970>
+
+2021-04-22 15:01:32 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Add interlace-mode back to caps for camera
+ skip_try_fmt_probes is set to TRUE for v4l2src to skip interlace-mode and
+ colorimetry when probe caps. gst_v4l2_object_set_format_full() will add
+ colorimetry back to caps when iterating over the negotiated caps. There is
+ one case that v4l2src is first in preview state then starts recording.
+ v4l2src caps will change with an additional interlace-mode structure after
+ renegotiation, then v4l2src needs to reset. But this camera driver can't
+ orphan buffer pool, it causes require buffer failed as streaming is still
+ in active state.
+ To fix this, also need to add interlace-mode back to caps for camera to
+ avoid reset.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/961>
+
+2021-04-02 18:41:28 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtpopuspay: set MARKER flag
+ Set MARKER flag on first buffer after DTX.
+ According to RFC 3551 section 4.1 the marker bit needs to be set on
+ "the first packet after a silence period during which packets have
+ not been transmitted contiguously".
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
+
+2021-03-31 11:18:30 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtpopuspay: add DTX support
+ If enabled, the payloader won't transmit empty frames.
+ Can be tested using:
+ opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
+
+2021-04-24 11:15:50 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * ext/taglib/gstid3v2mux.cc:
+ taglib: Update createFrame() to non-deprecated version.
+ ID3v2::FrameFactory::createFrame() versions not taking a Header have
+ been deprecated since v1.5 (Jan 2008).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/965>
+
+2021-04-25 02:16:45 +0200 Havard Graff <havard@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix divide-by-zero
+ The estimated packet-duration can sometimes end up as zero, and dividing
+ by that is never a good idea...
+ The test reproduces the scenario, and the fix is easy.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/966>
+
+2020-06-02 19:38:33 +0200 Havard Graff <havard@pexip.com>
+
+ rtpjitterbuffer: clean up and improve missing packets handling
+ * Try to make variable and function names more clear.
+ * Add plenty of comments describing the logic step-by-step.
+ * Improve the logging around this, making the logs easier to read and
+ understand when debugging these issues.
+ * Revise the logic of packets that are actually beyond saving in doing
+ the following:
+ 1. Do an optimistic estimation of which packets can still arrive.
+ 2. Based on this, find which packets (and duration) are now hopelessly
+ lost.
+ 3. Issue an immediate lost-event for the hopelessly lost and then add
+ lost/rtx timers for the ones we still hope to save, meaning that if
+ they are to arrive, they will not be discarded.
+ * Revise the use of rtx-delay:
+ Earlier the rtx-delay would vary, depending on the pts of the latest
+ packet and the estimated pts of the packet it being issued a RTX for,
+ but now that we aim to estimate the PTS of the missing packet accurately,
+ the RTX delay should remain the same for all packets.
+ Meaning: If the packet have a PTS of X, the delay in asked for a RTX
+ for this packet is always a constant X + delay, not a variable one.
+ * Finally ensure that the chaotic "check-for-stall" tests uses timestamps
+ that starts from 0 to make them easier to debug.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/952>
+
+2021-04-23 12:07:52 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * gst/level/gstlevel.c:
+ * gst/level/gstlevel.h:
+ level: make properties thread-safe
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>
+
+2021-04-22 14:11:09 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * gst/level/gstlevel.c:
+ level: disable passthrough when audio-level-meta is enabled
+ Ensure we receive a writable buffer to add the meta.
+ Fix #878
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>
+
+2021-04-23 08:28:06 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Don't pass a non-GObject pointer to GST_DEBUG_OBJECT and similar
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/964>
+
+2021-04-22 08:57:23 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Avoid generation of invalid timestamps
+ When updating timestamps and timer timeouts with a new offset, make sure that
+ the resulting value is valid (and not a negative (signed) value which ends up in
+ a massive (unsigned) value).
+ Fixes #571
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/960>
+
+2021-04-21 18:41:08 +0100 Philippe Normand <philn@igalia.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Fix glib warning emitted when attribute query fails
+ The v4l2object is not a GstObject. Logging has to go through its dbg_obj
+ specially meant for this.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/959>
+
+2021-03-25 13:20:38 +0100 VaL Doroshchuk <valentyn.doroshchuk@qt.io>
+
+ * ext/qt/gstqtoverlay.cc:
+ * tests/examples/qt/qmloverlay/overlay.py:
+ qmloverlay: Use first found GstGLVideoItem as widget property
+ GstGLVideoItem is required to render input video in the overlay's qml.
+ And currently qmlgloverlay requires to set this GstGLVideoItem to its widget property.
+ Instead of fetching GstGLVideoItem from the overlay's root object (root-item prop),
+ and setting it back as a widget (widget prop),
+ proposing to use found GstGLVideoItem in the current object hierarchy (passed in qml-scene) by default.
+ Also useful in Python, which solves the issue when casting gpointer <=> QQuickItem* is required.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/919>
+
+2021-04-19 16:39:03 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2.c:
+ v4l2: fix debug category initialisation again
+ Would spew warnings on the rpi4 when calling into
+ gst_v4l2_object_get_codec_caps() from the probe_and_register()
+ function since the v4l2_debug category initialisation would
+ only be done later as part of the element/device provider
+ registration.
+ Also log things in the probe function to the v4l2 category
+ instead of the default category while we're at it.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/956>
+
+2021-04-19 01:29:33 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix race saving seek event seqnum.
+ We need to save the seek seqnum before the flush stop event
+ since that will start the basesrc task which may send the segment
+ event before we're ready.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/954>
+
+2021-03-31 10:52:14 +0200 Marco Felsch <m.felsch@pengutronix.de>
+
+ * ext/qt/qtitem.cc:
+ * ext/qt/qtitem.h:
+ qmlglsink: allow to set force-aspect-ratio property
+ Add the forceAspectRatio Q_PROPERTY to allow changing the aspect ratio
+ from QML code as well.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/937>
+
+2021-04-19 11:14:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2src: fix spurious SOURCE_CHANGED error-level log messages
+ They're harmless, and some drivers at least return EINVAL
+ instead of ENOTTY for unsupported events (here: uvcvideo).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/955>
+
+2021-04-14 16:32:06 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtp/gstrtpsbcpay.c:
+ rtpsbcpay: remove use of packed struct for payload
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/950>
+
+2021-04-14 11:13:45 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/dtmf/gstdtmfcommon.h:
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ dtmf: convert to bit accessors
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/949>
+
+2021-04-13 09:23:12 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Remove some dead code
+ stop is not used after this point, nor do we create a new segment
+ here since 84725d62b57bc74ce34abde755f35bf8f948f94d
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>
+
+2021-04-10 02:53:51 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Do not overwrite the known duration after a seek
+ This breaks the duration query and also the seeking query.
+ Broke in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>
+
+2021-04-10 04:40:46 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Just assign the segment instead of memcpy
+ Assignments copy by value, we don't need to memcpy...
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>
+
+2021-04-13 11:30:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Check srcresult before waiting on the condition variable too
+ It might've been set to FLUSHING between the last check and the waiting,
+ and in that case we'd be waiting here forever now.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/944>
+
+2021-04-12 23:15:17 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * tests/check/elements/rtpsession.c:
+ rtp: fix test_twcc_header_and_run to support big endian.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>
+
+2021-04-12 23:13:15 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtpmanager/rtptwcc.c:
+ rtp: fix rtptwcc to support big endian.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>
+
+2021-04-12 21:59:45 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * gst/rtpmanager/gstrtphdrext-rfc6464.c:
+ rtp: fix rtphdrextrfc6464 to support big endian.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>
+
+2021-04-12 21:36:58 -0400 Doug Nazar <nazard@nazar.ca>
+
+ * tests/check/elements/alpha.c:
+ tests: Fix alpha test on big endian machines.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/943>
+
+2021-03-19 02:51:20 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Protect against writing absurd sample durations
+ If the input DTS goes backward or is missing, the calculated
+ sample duration goes negative and wraps around to a very big
+ number. In that case, just write a sample with a duration of
+ 0 and hope the problem is transient.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/914>
+
+2021-04-10 03:09:44 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: De-dup seek event seqnums to avoid multiple seeks
+ Seek events are sent upstream on each sink, so if we receive multiple
+ seeks with the same seqnum, we must only perform one seek, not N seeks
+ where N = the number of sinks in the pipeline connected to rtspsrc.
+ This is the same thing done by demuxers like qtdemux or matrsokademux.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/938>
+
+2021-04-10 01:55:28 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Using multicast UDP has no relation to seekability
+ The transport has no relation to whether a media can be seeked. The
+ range response having a duration is the correct thing to check for.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>
+
+2021-04-10 01:54:48 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Add more logging for range parsing and seekable
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>
+
+2021-04-10 14:47:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ videocrop: Update documentation cache
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-07 21:57:11 +0200 Markus Ebner <info@ebner-markus.de>
+
+ * gst/videocrop/gstvideocrop-private.h:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videocrop/gstvideocrop.h:
+ videocrop: Add support for GBR* video formats
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-07 18:54:49 +0200 Markus Ebner <info@ebner-markus.de>
+
+ * gst/videocrop/gstvideocrop-private.h:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videocrop/gstvideocrop.h:
+ videocrop: Added support for planar pixel formats > 8bits
+ - Added support for planar pixel formats with depths greater than 8bits
+ to transform_planar implementation
+ - Added a whole lot of new pixel formats to the support-list
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-07 17:52:34 +0200 Markus Ebner <info@ebner-markus.de>
+
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstvideocrop-private.h:
+ * gst/videocrop/gstvideocrop.c:
+ videocrop: Move supported format list into private header
+ - Moved declaration of supported pixel formats to private header, which
+ can be shared between videocrop and aspectvideocrop
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-06 17:02:34 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: More logging when calculating rfc7273 timestamps
+ This code can be fragile, since it is very exacting in the timestamps
+ that it will accept. Add more logging so it's easier to debug issues
+ and figure out whether it's a bug in the calculation or something
+ wrong in the incoming buffers.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/934>
+
+2021-04-08 13:29:10 +0200 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ rtp: missing debug init after element splitting
+ - h264depay
+ - h265depay
+ - sv3vdepay
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/936>
+
+2020-03-30 09:29:07 +0200 Michal Dzik <michal.dzik@streamunlimited.com>
+
+ * gst/rtp/gstrtpsbcpay.c:
+ rtp: rename gst_rtp_sbc_pay_flush_buffers()
+ gst_rtp_sbc_pay_flush_buffers() is a misleading name. A better name would
+ be gst_rtp_sbc_pay_drain_buffers(), because that's what it does, it drains
+ any leftover queued data and pushes it downstream. "Flushing" in GStreamer
+ typically means to throw away any queued data and not process/push it
+ downstream.
+ Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>
+
+2020-03-24 13:31:00 +0100 Michal Dzik <michal.dzik@streamunlimited.com>
+
+ * gst/rtp/gstrtpsbcpay.c:
+ rtp: fix adapter flushing in sbc payloader
+ GstAdapter must be flushed in some cases (flush, new segment, state change)
+ Without it, it may, for example, push some leftover buffer from old
+ segment in new segment. This, in general, breaks timestamps.
+ See GstAdapter documentation for more.
+ Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>
+
+2020-08-18 20:16:06 +0200 Jakub Adam <jakub.adam@collabora.com>
+
+ * ext/vpx/gstvpxenc.c:
+ vpxenc: add colorspace information into VP9 bitstream
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/871>
+
+2021-03-26 16:26:22 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Use default colorimetry if that in caps is unknown
+ Some streams have unknown colorimetry in caps, but v4l2object sets
+ default values for each primaries. It will cause check colorimetry
+ fail when do gst_v4l2_video_colorimetry_matches().
+ To fix this, need to keep the unknown colorimetry in caps same as
+ the default value set by v4l2object.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/923>
+
+2021-03-31 16:37:56 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Take segment stop into account when need_segment
+ Otherwise, in the case of e.g. a deferred seek event, the segment stop
+ would be replaced with GST_CLOCK_TIME_NONE.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/929>
+
+2021-03-29 16:45:26 +0200 Val Doroshchuk <val@sevendof.com>
+
+ * ext/qt/gstqtoverlay.cc:
+ * ext/qt/gstqtoverlay.h:
+ gstqtoverlay: Add initialization and finalization to qml-scene prop
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/924>
+
+2021-03-31 10:21:59 +1100 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/gstqtglutility.h:
+ qt: fix build warning with clang and c-linkage of user defined type
+ In file included from ../subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:17:
+ ../subprojects/gst-plugins-good/ext/qt/gstqtglutility.h:35:16: error: 'qt_opengl_native_context_from_gst_gl_context' has C-linkage specified, but returns user-defined type 'QVariant' which is incompatible with C [-Werror,-Wreturn-type-c-linkage]
+ QVariant qt_opengl_native_context_from_gst_gl_context (GstGLContext * context);
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/928>
+
+2021-03-30 09:45:45 +0200 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/qt/gstqtelement.cc:
+ * ext/qt/gstqtelements.h:
+ * ext/qt/gstqtoverlay.cc:
+ * ext/qt/gstqtsink.cc:
+ * ext/qt/gstqtsrc.cc:
+ qt: hotfix: allow per feature registration
+ Fixes #869
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/925>
+
+2021-02-17 08:52:40 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/twolame/gsttwolamemp2enc.c:
+ * ext/twolame/gsttwolamemp2enc.h:
+ twolame: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:49:03 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/shout2/gstshout2.c:
+ * ext/shout2/gstshout2.h:
+ shout2: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:38:46 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/cairo/gstcairo.c:
+ * ext/cairo/gstcairooverlay.c:
+ * ext/cairo/gstcairooverlay.h:
+ cairo: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:34:34 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/y4m/gsty4mencode.c:
+ * gst/y4m/gsty4mencode.h:
+ y4m: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:32:26 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/wavparse/gstwavparse.c:
+ * gst/wavparse/gstwavparse.h:
+ wavparse: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:29:40 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavenc/gstwavenc.h:
+ wavenc: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:27:24 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/spectrum/gstspectrum.c:
+ * gst/spectrum/gstspectrum.h:
+ spectrum: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:27:12 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/monoscope/gstmonoscope.c:
+ * gst/monoscope/gstmonoscope.h:
+ monoscope: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:22:47 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/imagefreeze/gstimagefreeze.h:
+ imagefreeze: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:19:52 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/id3demux/gstid3demux.c:
+ * gst/id3demux/gstid3demux.h:
+ id3demux: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:16:33 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/icydemux/gsticydemux.c:
+ * gst/icydemux/gsticydemux.h:
+ icydemux: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:14:26 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:11:26 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/cutter/gstcutter.c:
+ * gst/cutter/gstcutter.h:
+ cutter: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:11:14 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/goom/gstgoom.c:
+ * gst/goom/gstgoom.h:
+ goom: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:10:33 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlace.h:
+ deinterlace: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 16:34:48 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * sys/oss4/gstoss4audioplugin.c:
+ * sys/oss4/meson.build:
+ * sys/oss4/oss4-audio.c:
+ * sys/oss4/oss4-audio.h:
+ * sys/oss4/oss4-sink.c:
+ * sys/oss4/oss4-sink.h:
+ * sys/oss4/oss4-source.c:
+ * sys/oss4/oss4-source.h:
+ oss4: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 16:11:36 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * sys/oss/gstossaudio.c:
+ * sys/oss/gstossaudioelement.c:
+ * sys/oss/gstossaudioelements.h:
+ * sys/oss/gstosssink.c:
+ * sys/oss/gstosssrc.c:
+ * sys/oss/meson.build:
+ oss: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 15:56:35 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/auparse/gstauparse.c:
+ * gst/auparse/gstauparse.h:
+ auparse: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 15:29:06 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * sys/v4l2/gstv4l2.c:
+ * sys/v4l2/gstv4l2deviceprovider.c:
+ * sys/v4l2/gstv4l2element.c:
+ * sys/v4l2/gstv4l2elements.h:
+ * sys/v4l2/gstv4l2radio.c:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/meson.build:
+ v4l2: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 15:05:43 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/videofilter/gstgamma.c:
+ * gst/videofilter/gstgamma.h:
+ * gst/videofilter/gstvideobalance.c:
+ * gst/videofilter/gstvideobalance.h:
+ * gst/videofilter/gstvideoflip.c:
+ * gst/videofilter/gstvideoflip.h:
+ * gst/videofilter/gstvideomedian.c:
+ * gst/videofilter/gstvideomedian.h:
+ * gst/videofilter/plugin.c:
+ videofilter: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:58:57 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videocrop/gstvideocropelement.c:
+ * gst/videocrop/gstvideocropelements.h:
+ * gst/videocrop/gstvideocropplugin.c:
+ * gst/videocrop/meson.build:
+ videocrop: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:54:15 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/videobox/gstvideobox.c:
+ * gst/videobox/gstvideobox.h:
+ videobox: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:49:56 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/udp/gstdynudpsink.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudp.c:
+ * gst/udp/gstudpelement.c:
+ * gst/udp/gstudpelements.h:
+ * gst/udp/gstudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/meson.build:
+ udp: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:43:32 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/smpte/gstsmpte.c:
+ * gst/smpte/gstsmpte.h:
+ * gst/smpte/gstsmptealpha.c:
+ * gst/smpte/gstsmptealpha.h:
+ * gst/smpte/plugin.c:
+ smpte: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:38:37 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/shapewipe/gstshapewipe.c:
+ * gst/shapewipe/gstshapewipe.h:
+ shapewipe: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:35:51 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/rtsp/gstrtpdec.c:
+ * gst/rtsp/gstrtsp.c:
+ * gst/rtsp/gstrtspelement.c:
+ * gst/rtsp/gstrtspelements.h:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/meson.build:
+ rtsp: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:24:33 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpdtmfmux.c:
+ * gst/rtpmanager/gstrtpdtmfmux.h:
+ * gst/rtpmanager/gstrtpfunnel.c:
+ * gst/rtpmanager/gstrtpfunnel.h:
+ * gst/rtpmanager/gstrtphdrext-rfc6464.c:
+ * gst/rtpmanager/gstrtphdrext-rfc6464.h:
+ * gst/rtpmanager/gstrtphdrext-twcc.c:
+ * gst/rtpmanager/gstrtphdrext-twcc.h:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.h:
+ * gst/rtpmanager/gstrtpmanager.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ * gst/rtpmanager/gstrtpptdemux.c:
+ * gst/rtpmanager/gstrtpptdemux.h:
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxqueue.h:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/gstrtprtxreceive.h:
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/gstrtprtxsend.h:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/gstrtpsession.h:
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ * gst/rtpmanager/gstrtpssrcdemux.h:
+ * gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+ * gst/rtpmanager/gstrtpst2022-1-fecdec.h:
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.h:
+ rtpmanager: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 13:49:15 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/replaygain/gstrganalysis.c:
+ * gst/replaygain/gstrglimiter.c:
+ * gst/replaygain/gstrglimiter.h:
+ * gst/replaygain/gstrgvolume.c:
+ * gst/replaygain/gstrgvolume.h:
+ * gst/replaygain/replaygain.c:
+ * gst/replaygain/rganalysis.h:
+ replaygain: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 13:43:44 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/multipart/multipart.c:
+ * gst/multipart/multipartdemux.c:
+ * gst/multipart/multipartdemux.h:
+ * gst/multipart/multipartmux.c:
+ * gst/multipart/multipartmux.h:
+ multipart: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 12:04:26 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/multifile/gstimagesequencesrc.c:
+ * gst/multifile/gstimagesequencesrc.h:
+ * gst/multifile/gstmultifile.c:
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesink.h:
+ * gst/multifile/gstmultifilesrc.c:
+ * gst/multifile/gstmultifilesrc.h:
+ * gst/multifile/gstsplitfilesrc.c:
+ * gst/multifile/gstsplitfilesrc.h:
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/multifile/gstsplitmuxsrc.h:
+ multifile: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 11:14:17 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/matroska/gstmatroskaelement.c:
+ * gst/matroska/gstmatroskaelements.h:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/matroska-parse.h:
+ * gst/matroska/matroska.c:
+ * gst/matroska/meson.build:
+ * gst/matroska/webm-mux.c:
+ matroska: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 10:59:34 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/level/gstlevel.c:
+ * gst/level/gstlevel.h:
+ level: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 10:57:58 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/law/alaw-decode.c:
+ * gst/law/alaw-decode.h:
+ * gst/law/alaw-encode.c:
+ * gst/law/alaw-encode.h:
+ * gst/law/alaw.c:
+ * gst/law/mulaw-decode.c:
+ * gst/law/mulaw-decode.h:
+ * gst/law/mulaw-encode.c:
+ * gst/law/mulaw-encode.h:
+ * gst/law/mulaw.c:
+ law: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 10:26:40 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/isomp4/gstisomp4element.c:
+ * gst/isomp4/gstisomp4elements.h:
+ * gst/isomp4/gstqtmoovrecover.c:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstrtpxqtdepay.c:
+ * gst/isomp4/isomp4-plugin.c:
+ * gst/isomp4/meson.build:
+ * gst/isomp4/qtdemux.c:
+ isomp4: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 09:57:27 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/interleave/deinterleave.c:
+ * gst/interleave/gstinterleaveelements.h:
+ * gst/interleave/interleave.c:
+ * gst/interleave/plugin.c:
+ interleave: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 09:51:16 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/flx/gstflxdec.c:
+ * gst/flx/gstflxdec.h:
+ flx: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 17:37:09 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvelement.c:
+ * gst/flv/gstflvelements.h:
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvplugin.c:
+ * gst/flv/meson.build:
+ flv: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 17:27:51 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/equalizer/gstiirequalizer.c:
+ * gst/equalizer/gstiirequalizer.h:
+ * gst/equalizer/gstiirequalizer10bands.c:
+ * gst/equalizer/gstiirequalizer3bands.c:
+ * gst/equalizer/gstiirequalizernbands.c:
+ * gst/equalizer/gstiirequalizerplugin.c:
+ * gst/equalizer/meson.build:
+ equalizer: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 15:37:52 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/effectv/gstaging.c:
+ * gst/effectv/gstdice.c:
+ * gst/effectv/gstedge.c:
+ * gst/effectv/gsteffectv.c:
+ * gst/effectv/gsteffectv.h:
+ * 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:
+ effectv: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 15:03:10 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/dtmf/gstdtmf.c:
+ * gst/dtmf/gstdtmfsrc.c:
+ * gst/dtmf/gstdtmfsrc.h:
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ * gst/dtmf/gstrtpdtmfdepay.h:
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ * gst/dtmf/gstrtpdtmfsrc.h:
+ dtmf: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 14:55:15 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/debugutils/breakmydata.c:
+ * gst/debugutils/cpureport.c:
+ * gst/debugutils/gstcapsdebug.c:
+ * gst/debugutils/gstcapssetter.c:
+ * gst/debugutils/gstdebug.c:
+ * gst/debugutils/gstdebugutilselements.h:
+ * gst/debugutils/gstnavigationtest.c:
+ * gst/debugutils/gstnavigationtest.h:
+ * gst/debugutils/gstnavseek.c:
+ * gst/debugutils/gstpushfilesrc.c:
+ * gst/debugutils/gsttaginject.c:
+ * gst/debugutils/progressreport.c:
+ * gst/debugutils/rndbuffersize.c:
+ * gst/debugutils/testplugin.c:
+ debugutils: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 13:38:21 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/avi/gstavi.c:
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavielement.c:
+ * gst/avi/gstavielements.h:
+ * gst/avi/gstavimux.c:
+ * gst/avi/gstavisubtitle.c:
+ * gst/avi/meson.build:
+ avi: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 13:02:59 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/autodetect/gstautoaudiosink.c:
+ * gst/autodetect/gstautoaudiosrc.c:
+ * gst/autodetect/gstautodetect.c:
+ * gst/autodetect/gstautodetect.h:
+ * gst/autodetect/gstautodetectelement.c:
+ * gst/autodetect/gstautodetectelements.h:
+ * gst/autodetect/gstautodetectplugin.c:
+ * gst/autodetect/gstautovideosink.c:
+ * gst/autodetect/gstautovideosrc.c:
+ * gst/autodetect/meson.build:
+ autodetect: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 13:00:38 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstaudioparserselements.h:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ * gst/audioparsers/plugin.c:
+ audioparsers: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 12:44:31 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/apetag/gstapedemux.c:
+ * gst/apetag/gstapedemux.h:
+ apetag: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 11:00:46 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxelement.c:
+ * ext/vpx/gstvpxelements.h:
+ * ext/vpx/meson.build:
+ * ext/vpx/plugin.c:
+ vpx: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 17:26:36 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/taglib/gstapev2mux.cc:
+ * ext/taglib/gstid3v2mux.cc:
+ * ext/taglib/gsttaglibelement.c:
+ * ext/taglib/gsttaglibelements.h:
+ * ext/taglib/gsttaglibplugin.c:
+ * ext/taglib/meson.build:
+ taglib: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 17:09:19 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/qt/gstplugin.cc:
+ * ext/qt/gstqtelement.cc:
+ * ext/qt/gstqtelements.h:
+ * ext/qt/gstqtoverlay.cc:
+ * ext/qt/gstqtsink.cc:
+ * ext/qt/gstqtsrc.cc:
+ * ext/qt/meson.build:
+ * ext/qt/qtplugin.pro:
+ qt: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 16:09:53 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/speex/gstspeex.c:
+ * ext/speex/gstspeexdec.c:
+ * ext/speex/gstspeexelement.c:
+ * ext/speex/gstspeexelements.h:
+ * ext/speex/gstspeexenc.c:
+ * ext/speex/meson.build:
+ speex: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 16:04:16 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/soup/gstsoup.c:
+ * ext/soup/gstsoupelement.c:
+ * ext/soup/gstsoupelements.h:
+ * ext/soup/gstsouphttpclientsink.c:
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/meson.build:
+ soup: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:53:19 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/raw1394/gst1394.c:
+ * ext/raw1394/gstdv1394src.c:
+ * ext/raw1394/gstdv1394src.h:
+ * ext/raw1394/gsthdv1394src.c:
+ * ext/raw1394/gsthdv1394src.h:
+ raw1394: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:47:46 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/wavpack/gstwavpack.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * ext/wavpack/gstwavpackelement.c:
+ * ext/wavpack/gstwavpackelements.h:
+ * ext/wavpack/gstwavpackenc.c:
+ * ext/wavpack/meson.build:
+ wavpack: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:35:11 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/alpha/gstalpha.c:
+ * gst/alpha/gstalpha.h:
+ alpha: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:27:31 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/audiofx/audioamplify.c:
+ * gst/audiofx/audioamplify.h:
+ * gst/audiofx/audiochebband.c:
+ * gst/audiofx/audiochebband.h:
+ * gst/audiofx/audiocheblimit.c:
+ * gst/audiofx/audiocheblimit.h:
+ * gst/audiofx/audiodynamic.c:
+ * gst/audiofx/audiodynamic.h:
+ * gst/audiofx/audioecho.c:
+ * gst/audiofx/audioecho.h:
+ * gst/audiofx/audiofirfilter.c:
+ * gst/audiofx/audiofirfilter.h:
+ * gst/audiofx/audiofx.c:
+ * gst/audiofx/audioiirfilter.c:
+ * gst/audiofx/audioiirfilter.h:
+ * gst/audiofx/audioinvert.c:
+ * gst/audiofx/audioinvert.h:
+ * gst/audiofx/audiokaraoke.c:
+ * gst/audiofx/audiokaraoke.h:
+ * gst/audiofx/audiopanorama.c:
+ * gst/audiofx/audiopanorama.h:
+ * gst/audiofx/audiowsincband.c:
+ * gst/audiofx/audiowsincband.h:
+ * gst/audiofx/audiowsinclimit.c:
+ * gst/audiofx/audiowsinclimit.h:
+ * gst/audiofx/gstscaletempo.c:
+ * gst/audiofx/gstscaletempo.h:
+ * gst/audiofx/gststereo.c:
+ * gst/audiofx/gststereo.h:
+ audiofx: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 13:16:28 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/rtp/gstasteriskh263.c:
+ * gst/rtp/gstasteriskh263.h:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL16depay.h:
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpL16pay.h:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpL24depay.h:
+ * gst/rtp/gstrtpL24pay.c:
+ * gst/rtp/gstrtpL24pay.h:
+ * gst/rtp/gstrtpL8depay.c:
+ * gst/rtp/gstrtpL8depay.h:
+ * gst/rtp/gstrtpL8pay.c:
+ * gst/rtp/gstrtpL8pay.h:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpac3depay.h:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpac3pay.h:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpamrdepay.h:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtp/gstrtpamrpay.h:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpbvdepay.h:
+ * gst/rtp/gstrtpbvpay.c:
+ * gst/rtp/gstrtpbvpay.h:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltdepay.h:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpceltpay.h:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpdvdepay.h:
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtpdvpay.h:
+ * gst/rtp/gstrtpelement.c:
+ * gst/rtp/gstrtpelements.h:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg722depay.h:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtpg722pay.h:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg723depay.h:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg723pay.h:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726depay.h:
+ * gst/rtp/gstrtpg726pay.c:
+ * gst/rtp/gstrtpg726pay.h:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpg729depay.h:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpg729pay.h:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgsmdepay.h:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpgsmpay.h:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstdepay.h:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtpgstpay.h:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261depay.h:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph261pay.h:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263depay.h:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pay.h:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph263pdepay.h:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph263ppay.h:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264depay.h:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpilbcdepay.h:
+ * gst/rtp/gstrtpilbcpay.c:
+ * gst/rtp/gstrtpilbcpay.h:
+ * gst/rtp/gstrtpisacdepay.c:
+ * gst/rtp/gstrtpisacdepay.h:
+ * gst/rtp/gstrtpisacpay.c:
+ * gst/rtp/gstrtpisacpay.h:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kdepay.h:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpj2kpay.h:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegdepay.h:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpjpegpay.h:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvdepay.h:
+ * gst/rtp/gstrtpklvpay.c:
+ * gst/rtp/gstrtpklvpay.h:
+ * gst/rtp/gstrtpldacpay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp1sdepay.h:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.h:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmp2tpay.h:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4adepay.h:
+ * gst/rtp/gstrtpmp4apay.c:
+ * gst/rtp/gstrtpmp4apay.h:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gdepay.h:
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4gpay.h:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmp4vdepay.h:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmp4vpay.h:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmpadepay.h:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmpapay.h:
+ * gst/rtp/gstrtpmparobustdepay.c:
+ * gst/rtp/gstrtpmparobustdepay.h:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtpmpvdepay.h:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtpmpvpay.h:
+ * gst/rtp/gstrtpopusdepay.c:
+ * gst/rtp/gstrtpopusdepay.h:
+ * gst/rtp/gstrtpopuspay.c:
+ * gst/rtp/gstrtpopuspay.h:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmadepay.h:
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmapay.h:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtppcmudepay.h:
+ * gst/rtp/gstrtppcmupay.c:
+ * gst/rtp/gstrtppcmupay.h:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqcelpdepay.h:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpqdmdepay.h:
+ * gst/rtp/gstrtpreddec.c:
+ * gst/rtp/gstrtpredenc.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsbcdepay.h:
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpsbcpay.h:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpsirendepay.h:
+ * gst/rtp/gstrtpsirenpay.c:
+ * gst/rtp/gstrtpsirenpay.h:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpspeexdepay.h:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpspeexpay.h:
+ * gst/rtp/gstrtpstorage.c:
+ * gst/rtp/gstrtpstreamdepay.c:
+ * gst/rtp/gstrtpstreamdepay.h:
+ * gst/rtp/gstrtpstreampay.c:
+ * gst/rtp/gstrtpstreampay.h:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtpsv3vdepay.h:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtptheoradepay.h:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtptheorapay.h:
+ * gst/rtp/gstrtpulpfecdec.c:
+ * gst/rtp/gstrtpulpfecenc.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbisdepay.h:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvorbispay.h:
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8depay.h:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvp8pay.h:
+ * gst/rtp/gstrtpvp9depay.c:
+ * gst/rtp/gstrtpvp9depay.h:
+ * gst/rtp/gstrtpvp9pay.c:
+ * gst/rtp/gstrtpvp9pay.h:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawdepay.h:
+ * gst/rtp/gstrtpvrawpay.c:
+ * gst/rtp/gstrtpvrawpay.h:
+ * gst/rtp/meson.build:
+ * tests/check/meson.build:
+ rtp: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 11:12:34 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/pulse/gstpulseelement.c:
+ * ext/pulse/gstpulseelements.h:
+ * ext/pulse/meson.build:
+ * ext/pulse/plugin.c:
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesrc.c:
+ pulse: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:41:29 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/mpg123/gstmpg123audiodec.c:
+ * ext/mpg123/gstmpg123audiodec.h:
+ mpeg123: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:33:50 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/libpng/gstpng.c:
+ * ext/libpng/gstpngdec.c:
+ * ext/libpng/gstpngdec.h:
+ * ext/libpng/gstpngenc.c:
+ * ext/libpng/gstpngenc.h:
+ libpng: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:27:18 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/lame/gstlamemp3enc.c:
+ * ext/lame/gstlamemp3enc.h:
+ * ext/lame/plugin.c:
+ lame: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:26:26 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/libcaca/gstcacaplugin.c:
+ * ext/libcaca/gstcacasink.c:
+ * ext/libcaca/gstcacasink.h:
+ * ext/libcaca/gstcacatv.c:
+ * ext/libcaca/gstcacatv.h:
+ * ext/libcaca/meson.build:
+ libcaca: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:09:46 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/jpeg/gstjpeg.c:
+ * ext/jpeg/gstjpegdec.c:
+ * ext/jpeg/gstjpegelements.h:
+ * ext/jpeg/gstjpegenc.c:
+ * ext/jpeg/gstjpegplugin.c:
+ * ext/jpeg/gstsmokedec.c:
+ * ext/jpeg/gstsmokeenc.c:
+ * ext/jpeg/meson.build:
+ jpeg: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 09:56:36 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/jack/gstjack.c:
+ * ext/jack/gstjack.h:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosrc.c:
+ jack: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 08:57:55 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * ext/gdk_pixbuf/gstgdkpixbufelement.c:
+ * ext/gdk_pixbuf/gstgdkpixbufelements.h:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufplugin.c:
+ * ext/gdk_pixbuf/gstgdkpixbufsink.c:
+ * ext/gdk_pixbuf/meson.build:
+ gdk_pixbuf: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 08:48:21 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/gtk/gstgtkglsink.c:
+ * ext/gtk/gstgtkglsink.h:
+ * ext/gtk/gstgtksink.c:
+ * ext/gtk/gstgtksink.h:
+ * ext/gtk/gstplugin.c:
+ gtk: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-11 19:53:30 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/flac/gstflac.c:
+ * ext/flac/gstflacdec.c:
+ * ext/flac/gstflacelement.c:
+ * ext/flac/gstflacelements.h:
+ * ext/flac/gstflacenc.c:
+ * ext/flac/gstflactag.c:
+ * ext/flac/meson.build:
+ flac: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-11 18:57:03 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * ext/dv/gstdv.c:
+ * ext/dv/gstdvdec.c:
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvelement.c:
+ * ext/dv/gstdvelements.h:
+ * ext/dv/meson.build:
+ dv: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2020-08-14 15:27:31 -0400 Julian Bouzas <julian.bouzas@collabora.com>
+
+ * ext/aalib/gstaaplugin.c:
+ * ext/aalib/gstaasink.c:
+ * ext/aalib/gstaasink.h:
+ * ext/aalib/gstaatv.c:
+ * ext/aalib/gstaatv.h:
+ * ext/aalib/meson.build:
+ aalib: allow per feature registration
+ Split plugin into features including
+ dynamic types which can be indiviually
+ registered during a static build.
+ More details here:
+ https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-03-19 17:19:43 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videocrop/gstvideocrop.h:
+ * tests/check/elements/videocrop.c:
+ videocrop: handle non raw caps features
+ Currently, videocrop, only negotiates raw caps (system memory) because
+ it's the type of memory it can modify. Nonetheless, it's also possible
+ for the element to handle non-raw caps when only adding the crop meta
+ is possible, in other words, when downstream buffer pools expose the
+ crop API.
+ This patch enable non-raw caps negotiation. If downstream doesn't
+ expose crop API and negotiated caps are featured, the negotiation
+ fails.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/915>
+
+2021-03-19 10:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Don't special-case G_SIGNAL_RUN_CLEANUP stage in signal accumulators
+ All these signals don't run the class handler in the CLEANUP stage.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/913>
+
+2021-03-19 10:34:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/shout2/gstshout2.c:
+ shout2: Don't register signal without class handler with G_SIGNAL_RUN_CLEANUP
+ There is no class handler to run during the CLEANUP stage.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/913>
+
+2021-03-23 16:59:28 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Avoid colorimetry mismatch for streams with invalid colorimetry
+ video-info sets gst colorimetry to default value when colorimetry in caps
+ is unparsable or invalid. Then v4l2object uses this gst colorimetry to do
+ mapping with v4l2 colorimetry. This may cause colorimetry mismatch when
+ check mapped gst colorimetry with that read from caps directly.
+ To fix this, need to correct gst colorimetry as that parsed from video-info
+ when check gst_v4l2_video_colorimetry_matches().
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/917>
+
+2021-03-19 10:52:26 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Add support for hdr10 stream playback
+ Colorimetry of hdr10 video is bt2100-pq with transfer as
+ GST_VIDEO_TRANSFER_SMPTE2084. So map GST_VIDEO_TRANSFER_SMPTE2084
+ to V4L2_XFER_FUNC_SMPTE2084 to support hdr10 stream playback.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/888>
+
+2021-03-20 10:41:29 -0500 Sid Sethupathi <sid.sethupathi@gmail.com>
+
+ * gst/shapewipe/gstshapewipe.c:
+ shapewipe: fix broken link in docs
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/916>
+
+2021-03-18 17:42:02 +0000 Alba Mendez <me@alba.sh>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix more signals
+ Behaviour change in GLib causes select-stream signal to discard
+ the value returned by handlers. See !909 for more info.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/912>
+
+2021-03-18 19:52:53 +1100 Matthew Waters <matthew@centricular.com>
+
+ * ext/jack/gstjack.c:
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/pulse/pulsesink.h:
+ * ext/qt/gstqsgtexture.cc:
+ * ext/qt/gstqtglutility.cc:
+ * ext/qt/qtglrenderer.cc:
+ * ext/qt/qtitem.cc:
+ * ext/qt/qtwindow.cc:
+ * ext/vpx/gstvpxdec.c:
+ * ext/vpx/gstvpxenc.c:
+ * gst/audioparsers/gstac3parse.h:
+ * sys/rpicamsrc/gstrpicamsrc.c:
+ * sys/ximage/ximageutil.c:
+ gst: don't use volatile to mean atomic
+ volatile is not sufficient to provide atomic guarantees and real atomics
+ should be used instead. GCC 11 has started warning about using volatile
+ with atomic operations.
+ https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719
+ Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/911>
+
+2021-03-17 15:54:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtsp/gstrtspsrc.c:
+ Update docs cache and fix before-send signal doc syntax
+ The docs for before-send were missing because of this
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>
+
+2021-03-17 13:18:34 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix accumulation of before-send signal return values
+ Since glib 2.62, the accumulated return values in RUN_CLEANUP override the
+ accumulated return values in RUN_FIRST. Since:
+ 1. We have a default handler that always returns TRUE, and
+ 2. User handlers are only run in RUN_FIRST, and
+ 3. Our accumulator just takes the latest return value
+ We were discarding the return value from the user handler and always
+ sending messages even if the user handler said not to. See
+ https://gitlab.gnome.org/GNOME/glib/-/issues/2352 for more details.
+ This signal does not need RUN_CLEANUP or RUN_FIRST, so just change it
+ to RUN_LAST so that it's emitted exactly once and accumulated once.
+ With this fix, this signal can now be used to intercept PAUSE when
+ going to GST_STATE_NULL so that the server does a TEARDOWN (if
+ necessary) and not a PAUSE, which will confuse other RTSP clients when
+ playing shared media.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>
+
+2021-03-17 11:32:08 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ Revert unusable workaround for PAUSE being sent when going NULL
+ Directly setting rtspsrc to the NULL state before putting the pipeline
+ in the NULL state usually works, but it can cause a deadlock in some
+ cases, so it's not a reliable mechanism to fix this.
+ This reverts commit f37afdafff1fd0a339966116261f5cd0de53f5d1:
+ "rtspsrc: Fix state changes from PAUSED to PLAYING"
+ and commit 76d624b2df5594a82269b94dffe8766a372d059d:
+ "rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL"
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/908>
+
+2021-03-16 19:25:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix parsing of the mediaclk:direct= field
+ Due to an off-by-one when parsing the string, the most significant digit
+ or the clock offset was skipped when parsing the offset.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/907>
+
+2021-03-16 00:08:43 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix state changes from PAUSED to PLAYING
+ This was accidentally broken in the last commit that touched this
+ because I missed the fall-through in the case immediately above this.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/906>
+
+2021-03-04 13:05:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: Fix extraction of multichannel WavPack
+ The old code had a couple of issues that all lead to potential memory
+ safety bugs.
+ - Use a constant for the Wavpack4Header size instead of using sizeof.
+ It's written out into the data and not from the struct and who knows
+ what special alignment/padding requirements some C compilers have.
+ - gst_buffer_set_size() does not realloc the buffer when setting a
+ bigger size than allocated, it only allows growing up to the maximum
+ allocated size. Instead use a GstAdapter to collect all the blocks
+ and take out everything at once in the end.
+ - Check that enough data is actually available in the input and
+ otherwise handle it an error in all cases instead of silently
+ ignoring it.
+ Among other things this fixes out of bounds writes because the code
+ assumed gst_buffer_set_size() can grow the buffer and simply wrote after
+ the end of the buffer.
+ Thanks to Natalie Silvanovich for reporting.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/859
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/902>
+
+2021-03-03 11:31:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Initialize track context out parameter to NULL before parsing
+ Various error return paths don't set it to NULL and callers are only
+ checking if the pointer is NULL. As it's allocated on the stack this
+ usually contains random stack memory, and more often than not the memory
+ of a previously parsed track.
+ This then causes all kinds of memory corruptions further down the line.
+ Thanks to Natalie Silvanovich for reporting.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/858
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/902>
+
+2021-03-15 12:57:19 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL
+ This usually doesn't matter, but it is disruptive when streaming from
+ a shared media since it will pause all other clients when any client
+ exits.
+ This new behaviour is opt-in and should be safe because you need to
+ set the NULL state on rtspsrc directly, instead of just on the
+ pipeline. See the updated documentation for an explanation.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/901>
+
+2021-01-18 15:54:43 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: handle GST_VIDEO_TRANSFER_BT601
+ V4L2 makes no difference between the BT.601 and BT.709 transfer
+ functions [1], but GStreamer does since 1.18 [2].
+ Adapt gst_v4l2_object_get_colorspace() and
+ gst_v4l2_object_set_format_full().
+ [1] https://linuxtv.org/downloads/v4l-dvb-apis-new/userspace-api/v4l/colorspaces-details.html#colorspace-smpte-170m-v4l2-colorspace-smpte170m
+ [2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/724
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/856>
+
+2021-03-11 22:22:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix title of a few properties docstrings
+ GstRtspSrc -> GstRTSPSrc
+ This would have been noticed by the since checker, but those
+ properties were introduced prior to that.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/899>
+
+2021-03-07 21:25:01 +0000 Vladimir Menshakov <vladimir.menshakov@gmail.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/wavpack/gstwavpackdec.c:
+ * ext/wavpack/gstwavpackdec.h:
+ wavpackdec: Add floating point format support
+ This commit negotiate F32 audio format if MODE_FLOAT used in wavpack file.
+ Wavpack float mode is always in 32-bit IEEE format.
+ The following pipeline plays distorted audio if source file is encoded in float mode:
+ gst-launch-1.0 filesrc ... ! wavpackparse ! wavpackdec ! pulsesink
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/894>
+
+2021-03-04 16:40:06 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska: also support push-mode from seek events sent to the element
+ Otherwise sending seek events would fail to actually seek.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/891>
+
+2021-02-26 10:49:10 +0100 Marc Leeman <m.leeman@televic.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ gstrtspsrc: 551 should not result in an unhandled error
+ Some cameras (e.g. HikVision DS-2CD2732F-IS) return "551 Option
+ not supported" when a command is sent that is not implemented
+ (e.g. PAUSE). Instead; it should return "501 Not Implemented".
+ This is wrong, as previously, the camera did announce support for PAUSE
+ in the OPTIONS.
+ In this case, handle the 551 as if it was 501 to avoid throwing errors
+ to application level. */
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/885>
+
+2021-03-01 14:32:40 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Do not expose profiles/levels in vp8/vp9 template caps
+ Vp8/vp9 supported profiles/levels are listed in decoder sink caps, but
+ there is no parser for these two formats and the demuxers also don't have
+ these information. It causes negotiation fail between demuxers and decoder
+ when check caps "accept = gst_caps_is_subset (caps, template_caps);".
+ To fix this, need to remove profiles/levels for vp8/vp9 formats in decoder
+ sink caps.
+ Fix #854
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/887>
+
+2021-03-03 18:30:39 +0900 Seungha Yang <seungha@centricular.com>
+
+ * gst/rtpmanager/gstrtphdrext-twcc.h:
+ rtpmanager: Fix an MSVC compile warning
+ We don't expect this object is a part of public library.
+ gstrtphdrext-twcc.c(45): warning C4273: 'gst_rtp_header_extension_twcc_get_type': inconsistent dll linkage
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/889>
+
+2021-02-24 13:25:43 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: fix src side frame rate negotiation
+ Negotiating v4l2h264dec ! v4l2h264enc transcoding pipelines fails in
+ case the encoder does not accept framerate=(fraction)0/1.
+ The acquired caps used for downstream negotiation are determined from
+ gst_v4l2_object_acquire_format(), which sets the GstVideoInfo::fps_n
+ and ::fps_d fields to 0.
+ To fix this, copy the frame rate from the sink side.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/882>
+
+2021-02-16 16:20:05 +0200 Jordan Petridis <jpetridis@gnome.org>
+
+ * sys/rpicamsrc/meson.build:
+ rpicamsrc: depend on posix threads and vchiq_arm
+ Could only test on rpi 3b+
+ Close #839
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/875>
+
+2021-02-11 14:48:07 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Silence traces around unsupported source change
+ Don't be too spamy about unsupported source change flags as these will be
+ commonly extended in the future.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-11 14:24:29 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Move preferred resolution query before the probe
+ As we lock the DV_TIMINGS (and standards in the future), we need to probe the
+ caps after, otherwise, we may endup fixating to an unsupported resolution,
+ which would lead to a not-negotiated error.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 16:37:01 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2src: Calculate framerate from DV timings
+ And use this framerate in our preference. Note that we also flush
+ the probed caps as it seems that the format enumeration may change
+ when a new source change event get triggered.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 15:52:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2rc: Add DV_TIMINGS query and locking
+ This adds support to DV_TIMINGS query and locking. The timing width and
+ height is then used as a preference.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 15:49:03 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Force renegotiation on resolution change
+ As mandated by the specification, make sure to cycle through streamoff
+ / streamon regardless if the caps have changed or not.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 14:52:14 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.h:
+ v4l2object: Remove unused streaming member
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 10:48:48 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Refactor to use PreferredCapsInfo structure
+ Avoid passing around a bare structure for the preference, this removes
+ the need to copy and free that structure and simplify the code. Also
+ fix a type in the structure name, Prefered -> Preferred.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:27:20 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Stub preferred resolution support
+ This stubs the ability to use preferred resolution from digital
+ video timings, analog TV standards or driver reported native
+ resolution.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-09 14:44:02 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Subscribe source_change for the current input
+ When we subscribe for source-change event, we need to specify for which
+ input. Make sure we subscribe for the current input.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:26:20 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2src: Add input signal status detection
+ As part of the support to select a preferred size, we can also
+ detect the signal status. This is a split patch so that feature
+ is separated to ease review.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:24:00 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Add helper to query input status
+ This is a wrapper around ENUM_INPUT renamed for readability.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:22:37 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2radio.c:
+ * sys/v4l2/gstv4l2tuner.c:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: Fix input/output index sign
+ This is an unsigned integer in the kernel API.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 16:59:44 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Add source resolution change support
+ This patch adds support for source resolution change detection.
+ Resolution change is signaled by drivers when a change in the detected
+ signal have been detected. This is notably seen on HDMI receivers.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 14:13:32 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ v4l2bufferpool: Handle resolution change event
+ This patch adds the detection, dequeuing and reporting of the SOURCE_CHANGE
+ event when the CH_RESOLUTION flag is set. The acquire function will now return
+ a new custom success called GST_V4L2_FLOW_RESOLUTION_CHANGE. In order to use
+ this new feature, elements must enable it by calling:
+ gst_v4l2_buffer_pool_enable_resolution_change (pool);
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 11:01:38 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/v4l2_calls.c:
+ v4l2object: Add event helpers
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 10:10:34 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: use FLOW_LAST_BUFFER
+ This uses the GST_V4L2_FLOW_LAST_BUFFER alias instead of
+ GST_FLOW_CUSTOM_SUCCESS to make the code more readable.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2018-12-10 14:10:05 +0100 Lucas Stach <l.stach@pengutronix.de>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: prefer NV12 over I420
+ Considering NV12 an 'odd' format is a historical artifact. This format
+ is now quite common, and usually preferable to I420 due to more memory
+ friendly access patterns.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/857>
+
+2021-02-18 10:34:25 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * gst/wavparse/gstwavparse.c:
+ * tests/check/elements/wavparse.c:
+ wavparse: fix seeking in READY state
+ wavparse claims to be able to support seeking in the READY state by
+ saving the pending seek event and actually seeking later after having parsed the
+ header.
+ Problem was that this seek event was reset on the READY to PAUSED
+ transition, making all this code useless. Fixing it by stop resetting
+ on READY to PAUSED transition as we already reset on PAUSED to READY
+ and when initiating the element.
+ Note that DTS marker detection isn't support in such scenario as
+ gst_type_find_helper_for_buffer() needs a buffer containing the
+ beginning of the stream.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
+
+2021-02-18 10:05:03 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * tests/check/elements/wavparse.c:
+ tests: wavparse: factor out create_pipeline()
+ No semantic change.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
+
+2021-02-18 00:34:02 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: update plugins cache with new h264 / vp8 depay properties
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2020-12-09 01:40:45 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264depay.h:
+ rtph264depay: expose request-keyframe property
+ When set, the depayloader will request new keyframes on packet
+ loss
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2020-12-09 01:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8depay.h:
+ rtpvp8depay: expose request-keyframe property
+ When set, the depayloader will request new keyframes on packet
+ loss
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2020-12-09 01:24:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264depay.h:
+ rtph264depay: expose wait-for-keyframe property
+ Similar to rtpvp8depay, when packet loss occurs, the depayloader
+ starts waiting for a keyframe.
+ We try to only stop waiting when all the packets for the new keyframe
+ have been received, by only resetting waiting_for_keyframe when
+ encountering the first packet of a keyframe, this is slightly
+ fragile because there is no bit that explicitly marks the start
+ of an access unit, so we rely on the existing picture_start
+ detection code.
+ As a consequence, the property is only meaningful when outputting
+ access units, and is ignored when outputting NALs directly.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2021-02-18 00:36:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/videomixer/videomixer2.c:
+ videomixer: document as deprecated
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/878>
+
+2021-02-16 22:20:17 +1100 Ashley Brighthope <ashley.b@reddegrees.com>
+
+ * gst/wavenc/gstwavenc.c:
+ wavenc: Fixed INFO chunk corruption, caused by odd sized data not being padded. Code style was updated.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/873>
+
+2020-12-07 19:51:35 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: add info regarding (non-standard) multichannel support
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-12-07 16:50:01 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: update plugins cache for rtpopus
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-12-01 20:09:58 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * tests/check/elements/rtpopus.c:
+ tests: add rtpopus multichannel test cases
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-12-01 16:43:32 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * gst/rtp/gstrtpopusdepay.c:
+ rtpopusdepay: support libwebrtc-compatible multichannel payload
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-11-30 21:49:48 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: support libwebrtc-compatible multichannel payload
+ When the audio has more than 2 channels, add optional fields to output
+ caps from which webrtcbin can generate SDP in the syntax recognized by
+ "multiopus" codec present in libwebrtc [1].
+ e.g. for 5.1 audio:
+ a=rtpmap:96 multiopus/48000/6
+ a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5
+ [1] https://webrtc-review.googlesource.com/c/src/+/129768
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-11-30 22:10:14 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * gst/rtp/gstrtpopuspay.c:
+ rtpopuspay: make use of gst_rtp_base_payload_set_outcaps_structure()
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2021-02-09 19:31:28 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/effectv/LICENSE:
+ effectv: Remove redundant license file
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/869>
+
+2021-02-05 00:55:12 +0000 Kevin Song <kevinbing.song@gmail.com>
+
+ * sys/v4l2/gstv4l2videoenc.c:
+ Apply 1 suggestion(s) to 1 file(s)
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
+
+2021-02-05 00:55:04 +0000 Kevin Song <kevinbing.song@gmail.com>
+
+ * sys/v4l2/gstv4l2videoenc.c:
+ Apply 1 suggestion(s) to 1 file(s)
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
+
+2021-02-04 13:43:17 +0800 Bing Song <bing.song@nxp.com>
+
+ * sys/v4l2/gstv4l2videoenc.c:
+ v4l2videoenc: support resolution change stream encode.
+ Resolution change stream transcoding will drain before send new video
+ frame buffer. Need encode video frame after process EOS.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
+
+2021-02-04 11:44:53 +0100 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: added support for cbcs encryption scheme
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/865>
+
+2021-01-21 18:04:58 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtpmanager/gstrtphdrext-rfc6464.c:
+ * gst/rtpmanager/gstrtphdrext-rfc6464.h:
+ * gst/rtpmanager/gstrtpmanager.c:
+ * gst/rtpmanager/meson.build:
+ * tests/check/elements/rtphdrextrfc6464.c:
+ * tests/check/meson.build:
+ rtp: add rtphdrextrfc6464
+ Header Extension for Client-to-Mixer Audio Level Indication as
+ defined in RFC 6464.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
+
+2020-06-16 12:01:30 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/level/gstlevel.c:
+ * gst/level/gstlevel.h:
+ * tests/check/elements/level.c:
+ level: add GstRTPAudioLevelMeta on buffers
+ This meta can be used by a RTP payloader to send the level information
+ to the peer.
+ Part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/446
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
+
+2021-02-03 17:10:20 +0200 Robert Swain <robert.swain@gmail.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Provide documentation for GST_DEINTERLACE_BUFFER_STATE
+ More information available in
+ https://gstconf.ubicast.tv/videos/interlacing-and-telecine-in-gstreamer/
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
+
+2021-01-30 16:16:13 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/deinterlace/gstdeinterlacemethod.c:
+ deinterlace: Fix telecine/onefield mixup
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/838
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
+
+2021-01-30 15:49:23 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/deinterlace/gstdeinterlacemethod.c:
+ deinterlace: Better alternate support
+ Improve line offset halving based on whether this field is top or
+ bottom.
+ Also handle the buffer state the same as mixed.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
+
+2021-01-14 01:12:06 +0800 Bing Song <bing.song@nxp.com>
+
+ * sys/v4l2/gstv4l2h265codec.c:
+ v4l2h265codec: fix HEVC profile string issue.
+ Keep HEVC profile compatible with other module.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/850>
+
+2020-12-15 10:41:40 +0800 Bing Song <bing.song@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2object: Need keep same transfer as input caps.
+ GST_VIDEO_TRANSFER_BT2020_12 and GST_VIDEO_TRANSFER_BT2020_10 will
+ be mapped to V4L2_XFER_FUNC_709. Need check input caps when map
+ V4L2_XFER_FUNC_709 back to GST_VIDEO_TRANSFER_BT2020_12 and
+ GST_VIDEO_TRANSFER_BT2020_10
+ Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/816
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/841>
+
+2020-12-07 10:01:53 +0100 Tobias Ronge <tobiasr@axis.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Do not wait for response while flushing
+ Due to the may_cancel flag in GstRTSPConnection, receiving might not get
+ cancelled when supposed to. In this case, gst_rtsp_src_receive_response
+ will have to wait until timeout instead but if busy receiving RTP
+ data, this timeout will never occur.
+ With this patch, gst_rtsp_src_receive_response returns GST_RTSP_EINTR
+ if flushing is set to TRUE instead of continuing to receive.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/831>
+
+2021-01-14 19:13:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/dv/meson.build:
+ meson: allow libdv subproject fallback
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/854>
+
+2020-12-21 13:55:58 +0100 Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Allow streams with no specified protection system ID
+ This is necessary in cases like CMAF where there won't be any events
+ passing thru.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/852>
+
+2021-01-07 16:57:27 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * docs/gst_plugins_cache.json:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Map correct video format for RGBA
+ Map V4L2_PIX_FMT_RGBA32 pixel format to GST_VIDEO_FORMAT_RGBA instead of
+ GST_VIDEO_FORMAT_RGB video format to support RGBA.
+ Fixes #823
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/848>
+
+2021-01-02 13:06:16 +0530 Sanchayan Maity <sanchayan@asymptotic.io>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Fix marker links
+ These should be with a single ':'. The double '::' results in a CI with
+ build failure message like below.
+ ERROR: [links]: (mandatory-link-not-found): Mandatory link Link GstSocketTimestamp -> None (GstSocketTimestamp) could not be resolved
+ ERROR: [check-missing-since-markers]: (missing-since-marker): Missing since marker for udpsrc:socket-timestamp
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>
+
+2020-12-17 11:24:07 +0530 Sanchayan Maity <sanchayan@asymptotic.io>
+
+ * docs/gst_plugins_cache.json:
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: Allow use of socket control message timestamps for DTS
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>
+
+2020-12-09 20:20:18 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/videofilter/gstvideoflip.c:
+ * gst/videofilter/gstvideoflip.h:
+ * tests/check/elements/videoflip.c:
+ videoflip: fix possible crash when setting the video-direction while running
+ A classic case of not enough locking.
+ One interesting thing with this is the interaction between the
+ rotation value and caps negotiation. i.e. the width/height of the caps
+ can be swapped depending on the video-direction property. We can't lock
+ the entirety of the caps negotiation for obvious reasons so we need to
+ do something else. This takes the approach of trying to use a single
+ rotation value throughout the entirety of the negotiation and then
+ subsequent output frame in a kind of latching sequence.
+ Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/792
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
+
+2020-12-09 19:49:47 +1100 Matthew Waters <matthew@centricular.com>
+
+ * tests/check/elements/videoflip.c:
+ * tests/check/meson.build:
+ tests: add tests for videoflip
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
+
+2020-12-30 13:38:46 +0100 Ignacio Casal Quinteiro <qignacio@amazon.com>
+
+ * gst/deinterlace/meson.build:
+ deinterlace: force -DPREFIX on macos
+ This is due to a bug in meson where it will not detect properly
+ the compiler if the symbols need an undercore.
+ https://github.com/mesonbuild/meson/issues/5482
+ Fixes #821
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/845>
+
+2020-12-15 11:36:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/842>
+
+2020-12-10 14:27:49 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Avoid deadlock when releasing a pad from a running muxer
+ Might not drain correctly
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/838>
+
+2020-12-11 11:24:14 +0800 Hou Qi <qi.hou@nxp.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Use active resolution during fallback colorspace probe
+ For legacy drivers that don't implement ENUM_FRAMESIZE, use active
+ resolution to probe colorspace. This can improve the accuracy of the
+ result when the colorspace depends on the resolution. This fixes a
+ wrong colorspace issue on board with vendor bsp at resolution 2560x1440.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/830>
+
+2020-12-12 04:02:37 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+ rtpst2022-1-fecdec: don't xor out of bounds
+ When reconstituting packets from a stream with variable packet
+ sizes, don't xor larger packets past the length of the protected
+ packet
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
+
+2020-12-12 04:00:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+ rtpst2022-1-fecenc: memset when reallocating xored payload
+ When protecting packets with a variable payload length, we
+ reallocate the xored payload when needed. It is a good idea
+ to memset the extended memory to 0 so that we don't xor
+ data with garbage!
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
+
+2020-12-12 03:56:11 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+ rtpst2022-1-fec-*: protect additional RTP header fields
+ While the standard is a bit vague about whether the padding,
+ extension and marker bits should be protected:
+ > The usage, by senders and receivers, of the following bits shall
+ > be defined by the associated video/audio transport standards:
+ It is obviously necessary and useful for some formats (eg VP8)
+ that those indeed be protected.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
+
+2020-12-12 03:28:56 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/elements/splitmuxsink.c:
+ splitmuxsink: Unit test - check format/opened/closed sequence
+ Check the sequence of format-location/fragment-opened/fragment-closed
+ events is respected. There should be 1 format-location call for each
+ fragment-opened message, and 1 fragment-closed for each.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-12-09 00:40:52 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Fix for 'reference bytes muxed' check.
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798
+ introduced a check in the need-new-fragment logic to avoid starting a
+ new fragment unless there has been some data on the reference stream,
+ but the check is done against the number of bytes that have been
+ received on the input, not the number that were released for output
+ into the current fragment.
+ Fix the check to remember and test against bytes that have been sent
+ for output.
+ This also fixes a problem where starting a new fragment fails to
+ request a new filename from the format-location signal.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-09-15 00:27:24 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Add debug for fragment opened/closed msgs
+ When posting fragment-opened and fragment-closed messages,
+ put a debug statement in the logs
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-08-18 16:06:14 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Convert asserts into element errors.
+ Change some g_assert into element errors so that they can be
+ caught and the pipeline shut down.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-07-10 15:36:54 +1000 Matthew Waters <matthew@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtpmanager/gstrtpfunnel.c:
+ * gst/rtpmanager/gstrtphdrext-twcc.c:
+ * gst/rtpmanager/gstrtphdrext-twcc.h:
+ * gst/rtpmanager/gstrtpmanager.c:
+ * gst/rtpmanager/meson.build:
+ rtpmanager: update for rtp header extensions
+ Provide an implementation of the transport-wide-cc header extension and
+ use it in rtpfunnel.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/808>
+
+2020-11-15 11:30:07 +0000 Jose Quaresma <quaresma.jose@gmail.com>
+
+ * sys/rpicamsrc/meson.build:
+ rpicamsrc: add vchostif library as it is required to build successful
+ fix: undefined reference to `vc_gencmd'
+ /usr/src/debug/gstreamer1.0-plugins-good/1.18.1-r0/build/../gst-plugins-good-1.18.1/sys/rpicamsrc/RaspiCamControl.c:1440: undefined reference to `vc_gencmd'
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/818>
+
+2020-11-25 17:51:24 +0100 Marijn Suijten <marijns95@gmail.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests/rtp-payloading: Use new AudioFormatInfo::fill_silence function
+ The function is renamed to be properly associated with AudioFormatInfo
+ (its instance) instead of AudioFormat (an unrelated enum), see [1] for
+ the rename itself.
+ [1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940
+
+2020-11-24 22:11:50 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/deinterlace/meson.build:
+ * meson.build:
+ deinterlace: Enable x86 assembly with nasm on MSVC
+ We need to remove x86inc.asm from the list of compiled assembly files
+ because it is not supposed to be compiled separately. It is directly
+ included by yadif.asm, and it exports no symbols.
+ The object file was getting ignored on all platforms except on msvc
+ where it was causing a linker hang when building with debugging
+ enabled because the object file had no debug symbols (or similar).
+ We've seen this before in FFmpeg too, which uses nasm:
+ https://gitlab.freedesktop.org/gstreamer/meson-ports/ffmpeg/-/merge_requests/46
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/825>
+
+2020-11-19 17:47:21 +1100 Matthew Waters <matthew@centricular.com>
+
+ * ext/qt/gstqtoverlay.cc:
+ * ext/qt/gstqtsink.cc:
+ qml: add some docs on display and contexts
+ Especially considering some dynamic pipeline scenarios.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/822>
+
+2020-11-18 20:09:24 +0100 Tim Schneider <tim.schneider94@t-online.de>
+
+ * sys/rpicamsrc/gstrpicamsrc.c:
+ rpicamsrc: Added "src->started = FALSE;" to gst_rpi_cam_src_stop
+ Makes the element reusable multiple times after a state change back to READY.
+ Fixes #105
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/823>
+
+2020-11-12 09:32:30 +0800 Bing Song <bing.song@nxp.com>
+
+ * docs/gst_plugins_cache.json:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: caps negotiate wrong as interlace feature
+ gst_caps_simplify() will move interlace format before normal video
+ format. It will cause caps negotiate prefer interlaced caps which
+ isn't expected. Seperate normal caps and interlaced caps and then
+ merge it will keep prefer progress video format.
+ Add ARGB/BGRA for interlaced caps.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/802
+ Part-of <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>
+
+2020-11-13 21:25:42 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: never send on a non-internal source
+ This will end up as a "received" packet, due to the code in
+ source_push_rtp, which will think this is a packet being received.
+ Instead drop the packet and hope that either:
+ 1. Something upstream responds to the GstRTPCollision event and changes
+ SSRC used for sending.
+ 2. That the application responds to the "on-ssrc-collision" signal, and
+ forces the sender (payloader) to change its SSRC.
+ 3. That the BYE sent to the existing user of this SSRC will respond to
+ the BYE, and that we timeout this source, so we can continue sending
+ using the chosen SSRC.
+ The test reproduces a scenario where we previously would have sent
+ on a non-internal source.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
+
+2020-11-13 12:39:53 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: rewrite timeout-check to avoid underflow
+ If current_time is < collision_timeout, we get an uint64 underflow, and
+ the check will trigger prematurely.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
+
+2020-11-13 14:58:44 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Fix caps change handling
+ In baseparse we set the fixed caps flag on all src pads, therefore the
+ source pad caps query in get_allowed_caps will return the current caps.
+ Current caps won't necessarily intersect with the new caps (e.g. sample
+ rate change). Replace get_allowed_caps with peer_query_caps.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/816>
+
+2020-11-12 23:39:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/qtdemux.c:
+ tests: qtdemux: fix typo in caps field
+ timesacle -> timescale
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
+
+2020-11-12 23:38:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/qtdemux.c:
+ tests: qtdemux: fix crash on 32-bit architectures
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/803
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
+
+2020-09-14 13:12:50 +0530 Sanchayan Maity <sanchayan@asymptotic.io>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpldacpay.c:
+ * gst/rtp/gstrtpldacpay.h:
+ * gst/rtp/meson.build:
+ rtp: ldacpay: Add LDAC RTP payloader
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/757>
+
+2020-11-03 15:58:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/qt/gstqsgtexture.cc:
+ * ext/qt/gstqsgtexture.h:
+ * ext/qt/qtitem.cc:
+ qmlglsink: Keep old buffers around a bit longer if they were bound by QML
+ We don't know exactly when QML will stop using them but it should be
+ safe to unref them after at least 2 more buffers were bound.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/810>
+
+2020-11-10 18:18:12 +0000 ChrisDuncanAnyvision <chrisd@anyvision.co>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Ensure same group-id used for both TCP/UDP stream-start events
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
+
+2020-11-10 16:17:23 +0000 ChrisDuncanAnyvision <chrisd@anyvision.co>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Use consistent URI hashed stream-id for UDP and TCP/Interleaved streams
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
+
+2020-11-04 18:43:04 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ meson: Enable some MSVC warnings for parity with GCC/Clang
+ This makes it easier to do development with MSVC by making it warn
+ on common issues that GCC/Clang error out for in our CI configuration.
+ Continuation from https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/223
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/809>
+
+2020-10-15 21:42:40 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.h:
+ rtpsource: Report for which local SSRC is a remote RB reporting on
+ This is useful in the Bundle case because there may be multiple local
+ and remote SSRCs in the same session.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/776>
+
+2020-10-29 15:58:38 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtp/gstrtpisacdepay.c:
+ * gst/rtp/gstrtpisacpay.c:
+ docs: update plugins cache
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
+
+2020-03-20 13:15:33 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpisacdepay.c:
+ * gst/rtp/gstrtpisacdepay.h:
+ * gst/rtp/meson.build:
+ rtp: add rtpisacdepay
+ Depayload for the iSAC audio codec.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
+
+2020-03-20 13:15:33 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpisacpay.c:
+ * gst/rtp/gstrtpisacpay.h:
+ * gst/rtp/meson.build:
+ rtp: add rtpisacpay
+ Payload for the iSAC audio codec.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
+
+2020-11-01 18:36:49 +0000 Dinesh Manajipet <saidinesh5@gmail.com>
+
+ * ext/qt/qtitem.cc:
+ qmlglsink: Set qtitem's implicit width/height
+ This can be useful to let the layouts automatically resize qtitem
+ and also easily query a video's width/height from QML
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/802>
+
+2020-11-01 10:30:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Release pads via GstAggregator
+ See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/801>
+
+2020-10-26 12:40:49 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: support muxing multiple codec_data for h264/h265
+ Each codec_data is put into its own SampleTableEntry inside the stsd.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/787>
+
+2020-10-29 14:54:16 +0100 Stéphane Cerveau <scerveau@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/debugutils/gstnavseek.c:
+ * gst/debugutils/gstnavseek.h:
+ navseek: add hold_eos property
+ This property will tell the element to hold
+ the EOS event and keep it until the next
+ keystroke.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/792>
+
+2020-10-31 12:52:04 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/elements/splitmuxsrc.c:
+ splitmuxsrc: Fix comment in a test
+ Fix a comment in the splitmuxsrc robust muxing test so it
+ describes the test properly.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-31 12:49:08 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Change EOS catching logic.
+ Add a new state for ending the overall stream, and use it to decide
+ whether to pass the final EOS message up the bus instead of dropping
+ it. Fixes a small race that makes the testsuite sometimes not generate
+ the last fragment(s) sometimes because the wrong EOS gets
+ allowed through too early.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-31 02:19:07 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Don't use the element state lock
+ Using the element state lock to avoid splitmuxsink shutting
+ down while doing element manipulations can lead to a deadlock on
+ shutdown if a fragment switch happens at exactly the wrong moment.
+ Use a private mutex and a shutdown boolean instead.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-30 03:38:15 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Don't busy loop on a non-ready pad.
+ If a pad gets into the check_completed_gop method and then
+ the underlying conditions change on the reference context,
+ things could get stuck in a busy loop when the context should
+ instead jump back out and wait for more data.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-30 03:36:51 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Mark running=false on shutdown.
+ Make sure that any late gst_element_call_async() callbacks
+ know that the elements is shutting down and bail out instead
+ of operating on the element we're trying to stop.
+ Fixes a spurious test failure in elements_splitmuxsrc
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-29 02:36:35 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Forward EOS messages from async fragments.
+ Re-enable forwarding EOS messages from fragments that are completing
+ asynchronously, so that splitmuxsink itself won't go EOS until they
+ are complete. This was disabled to work around a bug in core that
+ is fixed in
+ https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/683
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-09-17 22:56:01 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Never start a new fragment with no reference buffers
+ If there has been no bytes from the reference stream muxed into
+ the current fragment, then time can't have advanced, there's no
+ GOP... this fragment would be broken or empty, so wait for some
+ data on the reference buffer.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-29 02:38:16 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Chain up when releasing pad, and fix some locking.
+ Release pads by calling up into aggregator so it can do the right
+ things. Don't clean up the pad until after that.
+ Add some missing locks around some accesses to shared pad state.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797>
+
+2018-08-13 15:35:11 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ * gst/rtp/gstrtpvp9depay.h:
+ * tests/check/elements/rtpvp9.c:
+ rtpvp9depay: Improve SVC parsing, aggregate all layers
+ - Fix start and end of picture to support multiple layers. Start of
+ picture is the first packet of the base layer, while end of picture
+ is when the marker bit is set (last packet of the enhancement
+ layers).
+ - All "layers" (aka "frames") of a picture are pushed downstream in a
+ single buffer when picture is complete.
+ - Forgive SID=0 for enhancement layers (invalid, but Chrome and
+ Firefox sends it)
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/773>
+
+2020-10-30 03:09:48 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8depay.h:
+ * tests/check/elements/rtpvp8.c:
+ rtpvp8depay: Send lost events when marker bit is missing
+ This means the previous frame was incomplete.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/796>
+
+2020-10-14 23:17:53 +0200 Knut Saastad <Knut@bitflow.io>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp9depay: detect incomplete frames and bail out
+ If a packet with the B bit set arrives but we haven't received
+ a packet with the marker or E bits set to end the previous frame,
+ we know the current frame was incomplete.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/795>
+
+2020-10-14 23:17:53 +0200 Knut Saastad <Knut@bitflow.io>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ rtpvp9depay: detect incomplete frames and bail out
+ If a packet with the B bit set arrives but we haven't received
+ a packet with the marker or E bits set to end the previous frame,
+ we know the current frame was incomplete.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
+
+2020-10-14 01:28:50 +0200 Mikhail Fludkov <misha@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8depay.h:
+ * gst/rtp/gstrtpvp9depay.c:
+ * gst/rtp/gstrtpvp9depay.h:
+ * tests/check/elements/rtpvp8.c:
+ * tests/check/elements/rtpvp9.c:
+ rtpvp*depay: possibly forward might-have-been-fec PacketLost events
+ This is ad adaptation of a Pexip patch for dealing with spurious
+ GstRTPPacketLost events caused by lost ulpfec packets: as FEC packets
+ under that scheme are spliced in the same sequence domain as the media
+ packets, it is not generally possible to determine whether a lost packet
+ was a FEC packet or a media packet.
+ When upstreaming pexip's ulpfec patches, we decided to drop all lost
+ events at the base depayloader level, and where the original patch
+ from pexip was making use of picture ids and marker bits to determine
+ whether a packet should be forwarded, this patch makes use of those
+ to determine whether they should be dropped instead (by removing their
+ might-have-been-fec field).
+ Spurious lost events coming out of the depayloader can cause the
+ decoder to stop decoding until the next keyframe and / or request a new
+ keyframe, and while this is not desirable it makes sense to forward
+ that information when we have other means to determine whether a lost
+ packet was indeed a FEC packet, as is the case with VP8 / VP9 payloads
+ when they carry a picture id.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
+
+2020-10-20 23:22:36 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Preserve SPS/PPS arrival order.
+ Even if SPS/PPS haven't changed, make sure to move them to the
+ end of the tracking array if needed, so we always know what the
+ most recent entries are, in case we need to discard the oldest
+ when generating codec_data.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
+
+2020-10-17 00:05:15 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Warn when max SPS/PPS are collected in AVC mode.
+ The AVC codec_data has a flaw that it can only accomodate
+ 31 SPS headers, even though H.264 can have 32, and 255 PPS,
+ when there can be 256 in H.264. When streaming RTP some
+ clients like to cycle through SPS/PPS ids when changing
+ configuration and can eventually accumulate a full set.
+ In that case, we have no choice but to discard one (oldest)
+ entry, or else the count written into the codec_data is wrong
+ and downstream decoding failures ensue.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
+
+2020-10-28 00:29:05 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtptimerqueue.c:
+ * gst/rtpmanager/rtptimerqueue.h:
+ * tests/check/elements/rtpjitterbuffer.c:
+ * tests/check/elements/rtptimerqueue.c:
+ rtpjitterbuffer: don't send multiple instant RTX for the same packet
+ Due to us not properly acknowleding the time when the last RTX was sent
+ when scheduling a new one, it can easily happen that due to the packet
+ you are requesting have a PTS that is slightly old (but not too old when
+ adding the latency of the jitterbuffer), both its calculated second and
+ third (etc.) timeout could already have passed. This would lead to a burst
+ of RTX requests, which acts completely against its purpose, potentially
+ spending a lot more bandwidth than needed.
+ This has been properly reproduced in the test:
+ test_rtx_not_bursting_requests
+ The good news is that slightly re-thinking the logic concerning
+ re-requesting RTX, made it a lot simpler to understand, and allows us
+ to remove two members of the RtpTimer which no longer serves any purpose
+ due to the refactoring. If desirable the whole "delay" concept can actually
+ be removed completely from the timers, and simply just added to the timeout
+ by the caller of the API. But that can be a change for a another time.
+ The only external change (other than the improved behavior around bursting
+ RTX) is that the "delay" field now stricly represents the delay between
+ the PTS of the RTX-requested packet and the time it is requested on,
+ whereas before this calculation was more about the theoretical calculated
+ delay. This is visible in three other RTX-tests where the delay had
+ to be adjusted slightly. I am confident however that this change is
+ correct.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/789>
+
+2020-10-27 23:43:49 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Fix sparse stream crash
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656
+ introduced an invalid memory access when debug is enabled, by casting
+ the wrong pointer to a GstCollectPad. Fixing that showed the original
+ change was incorrect and leads to an infinite loop in the
+ testsuite. This patch fixes both problems.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/788>
+
+2020-10-22 15:29:01 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * ext/vpx/gstvpxenc.c:
+ vpx: Fix the check to unfixed/unknown framerate to set bitrate
+ 0/1 means unknown framerate not X/0 (which is illegal).
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/782>
+
+2020-10-22 09:17:26 -0400 Arun Raghavan <arun@asymptotic.io>
+
+ * gst/rtp/gstrtputils.c:
+ rtputils: Count metas with an empty tag list for copying/keeping
+ The GstMetaInfos registered in core do not set their tags to NULL, but
+ instead use an empty list (non-NULL list with a single NULL value).
+ Let's check explicitly for that so as to not miss some metas.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/779>
+
+2020-10-16 16:05:45 -0700 Bastien Reboulet <bastien.reboulet@gmail.com>
+
+ * ext/qt/qtitem.cc:
+ qmlglsink: fix crash when created/destroyed in quick succession
+ The crash is caused by a race condition where the render thread
+ calls a method on the QtGLVideoItem instance that was
+ previously destroyed by the main thread.
+ Also, less frequently, QtGLVideoItem::onSceneGraphInitialized
+ is called when QQuickItem::window is null, also causing a crash.
+ Fixes #798
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/777>
+
+2020-10-19 18:23:25 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ * sys/v4l2/gstv4l2videoenc.c:
+ v4l2codec: Garbage collect old frames if they accumulate because of codec bugs
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
+
+2020-10-19 17:56:04 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2sink.c:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2transform.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ * sys/v4l2/gstv4l2videoenc.c:
+ v4l2codec: Pass system frame number as timestamp and use it to retrieve back frames reliably
+ System frame numbers are supposed to be unique and correct drivers are
+ passing through timestamps without modification from the output/sink to the
+ capture/src side.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
+
+2020-09-24 13:13:00 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: Add clear-ssrc action
+ This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
+ associated with the session. This allow rtpbin users to clear pads and
+ elements for a specific ssrc that is known to no longer be in use. This
+ happens when a pad is reused in rtpsrc or ristsrc.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>
+
+2017-09-08 20:02:13 +0100 John-Mark Bell <jmb@pexip.com>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvp8pay.h:
+ * tests/check/elements/rtpvp8.c:
+ * tests/check/meson.build:
+ rtpvp8pay: payload temporally scaled bitstreams.
+ Co-Authored-By: Vincent Sanders <vince@pexip.com>
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-11-17 15:11:41 +0100 Stian Selnes <stian@pexip.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvp8pay.h:
+ rtpvp8pay: Add picture-id-offset property
+ Add property to set the initial value for picture-id. RFC7741 says
+ that picture-id MAY be initialized to a random value, thus it's also
+ valid to simply set it to a fixed initial value. A fixed value is very
+ useful for testing.
+ Default behavior is not changed.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-03-16 15:23:28 +0100 Mikhail Fludkov <misha@pexip.com>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: move duplicate code to separate functions
+ Two new functions to modify picture id:
+ gst_rtp_vp8_pay_picture_id_reset - picks random picture id of
+ appropriate bitsize
+ gst_rtp_vp8_pay_picture_id_increment - increments picture id taking
+ care of wrapping
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-09-08 08:13:05 +0100 John-Mark Bell <jmb@pexip.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/vpx/gstvpxenc.c:
+ vp8enc: expect bps for temporal-scalability-target-bitrate.
+ Consistency with target-bitrate is less surprising and with
+ modern libvpx additional configuration is required to make
+ temporal scaling work.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-09-08 08:19:20 +0100 John-Mark Bell <jmb@pexip.com>
+
+ vp8enc: finish support for temporally scaled encoding
+ - introduce two new properties:
+ * temporal-scalability-layer-flags:
+ Provide fine-grained control of layer encoding to the
+ outside world. The flags sequence should be a multiple of
+ the periodicity and is indexed by a running count of encoded
+ frames modulo the sequence length.
+ * temporal-scalability-layer-sync-flags:
+ Specify the pattern of inter-layer synchronisation (i.e.
+ which of the frames generated by the layer encoding
+ specification represent an inter-layer synchronisation).
+ There must be one entry per entry in
+ temporal-scalability-layer-flags.
+ - apply temporal scalability settings and expose as buffer
+ metadata.
+ This allows the codec to allocate a given frame to the correct
+ internal bitrate allocator. Additionally, all the
+ non-bitstream metadata needed to payload a temporally scaled
+ stream is now attached to each output buffer as a
+ GstVideoVP8Meta.
+ - add unit test for temporally scaled encoding.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2020-10-15 18:21:54 +0200 Stéphane Cerveau <scerveau@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/udp/gstudp.c:
+ * meson.build:
+ meson: update glib minimum version to 2.56
+ In order to support the symbol g_enum_to_string in various
+ project using GStreamer ( gst-validate etc.), the glib minimum
+ version should be 2.56.0.
+ Remove compat code as glib requirement
+ is now > 2.56
+ Version used by Ubuntu 18.04 LTS
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/774>
+
+2020-10-14 14:30:34 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+ rtpst2022-1-fecenc: fix input seqnum check
+ We need to cast the incremented last seqnum to guint16 for
+ consistent checks on wraparound
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/770>
+
+2020-09-12 09:02:30 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Correct time types
+ - last_dts is in milliseconds, not nanoseconds as expected for
+ GstClockTime. Make it a generic guint64.
+ - Use GstClockTime for the fields that actually contain nanoseconds.
+ None of them should become negative.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/766>
+
+2020-10-09 09:31:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+ rtpst2022-1-fecenc: Don't unconditionally use GLib 2.60 APIs
+ g_queue_clear_full() in this case.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/768>
+
+2020-10-08 18:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtp/rtpulpfeccommon.c:
+ rtpulpfec: fix potential alignment issue in xor function
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753#note_646453
+ for context
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
+
+2020-10-06 03:03:13 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpmanager.c:
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+ * gst/rtpmanager/gstrtpst2022-1-fecenc.h:
+ * gst/rtpmanager/meson.build:
+ * tests/check/elements/rtpst2022-1-fecenc.c:
+ * tests/check/meson.build:
+ rtpmanager: implement SMPTE 2022-1 FEC encoder
+ + improve integration of FEC encoders in rtpbin
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
+
+2020-10-06 03:13:30 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * gst/rtpmanager/gstrtpmanager.c:
+ * gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+ * gst/rtpmanager/gstrtpst2022-1-fecdec.h:
+ * gst/rtpmanager/meson.build:
+ * tests/check/elements/rtpst2022-1-fecdec.c:
+ * tests/check/meson.build:
+ rtpmanager: implement SMPTE 2022-1 FEC decoder
+ + improve integration of FEC decoders in rtpbin
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
+
+2020-07-08 17:28:31 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpfunnel.c:
+ * tests/check/elements/rtpfunnel.c:
+ rtpfunnel: Also forward custom sticky event
+ This is useful to track metadata about each group of packets
+ Also include a unit test
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/666>
+
+2020-09-29 09:44:54 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/isomp4/gstqtmux-doc.c:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ isomp4: Rename GstQTMux to GstBaseQTMux to avoid breaking API
+ Since 52b63de19ada283c1180c8fc00cacb1465fdf10f the qtmux GType was
+ renamed GstQTMuxElement which breaks presets, revert that change.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/755>
+
+2020-09-28 18:25:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvp9pay.c:
+ rtp: Fix allocations to support source-info property
+ Use gst_rtp_base_payload_allocate_output_buffer() instead of
+ gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with
+ correct number of CSRCs according to the meta.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
+
+2015-10-23 11:08:56 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: Fix allocation to support source-info property
+ Use gst_rtp_base_payload_allocate_output_buffer() in order to allocate
+ RTP buffer with correct number of CSRCs according to the meta.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/314
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
+
+2020-09-28 15:36:00 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: output the correct limits in error messages
+ Having the current bytes being less than the limit was confusing!
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
+
+2020-07-31 16:47:37 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ * tests/check/elements/qtmux.c:
+ qtmux: properly support initial caps nego failure
+ Scenario:
+ - gap event causes h264parse to push made up caps that may fail checks
+ inside qtmux (e.g missing codec_data).
+ - the caps event has already been marked as received and is sticky on
+ the sink pad
+ - gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event
+ using gst_pad_get_current_caps() and reject the correct updated caps
+ with codec_data.
+ - Failure!
+ Keep track of the configured caps ourselves instead of relying on the
+ sticky event on the pad.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
+
+2020-07-22 15:34:44 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: support non-seekable downstream mode
+ Write an mdat per buffer in that case.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
+
+2020-09-23 15:25:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Remove the rtpjitterbuffer with the stream
+ Since !348, the jitterbuffer was only removed with the session. This restores
+ the original behaviour and removes the jitterbuffer when the stream is
+ removed. This avoid accumulating jitterbuffer objects into the bin when a
+ session is reused.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
+
+2020-09-23 13:26:51 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Cleanup dead code
+ The rtpjitterbuffer is now part of the session elements, we no longer need
+ to do the ref_sink dance when signalling it. It is already owned by the bin
+ when signalled. Also, the code that handles generic session elements already
+ handle the ref_sink() calls since:
+ 03dc22951bacb6fdc3868c8f801e6a52c33a745f
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
+
+2020-09-18 16:09:20 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264depay.h:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * tests/check/elements/rtph264.c:
+ rtph26*depay: drop FU's without a corresponding start bit
+ If we have not received a FU with a start bit set, any subsequent FU
+ data is not useful at all and would result in an invalid stream.
+ This case is constructed from multiple requirements in
+ RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3. Following are excerpts
+ from RFC 3984 but RFC 7798 contains similar language.
+ The FU in a single FU case is forbidden:
+ A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the
+ Start bit and End bit MUST NOT both be set to one in the same FU
+ header.
+ and dropping is possible:
+ If a fragmentation unit is lost, the receiver SHOULD discard all
+ following fragmentation units in transmission order corresponding to
+ the same fragmented NAL unit.
+ The jump in seqnum case is supported by this from the specification
+ instead of implementing the forbidden_zero_bit mangling:
+ If a fragmentation unit is lost, the receiver SHOULD discard all
+ following fragmentation units in transmission order corresponding to
+ the same fragmented NAL unit.
+ A receiver in an endpoint or in a MANE MAY aggregate the first n-1
+ fragments of a NAL unit to an (incomplete) NAL unit, even if fragment
+ n of that NAL unit is not received. In this case, the
+ forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
+ syntax violation.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/730>
+
+2020-09-20 21:06:19 +0900 Seungha Yang <seungha@centricular.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Response caps query from srcpad
+ ... and chain up to default query handler for unhandled query types.
+ Unhandled query shouldn't be returned with FALSE if there's no special needs.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/731>
+
+2020-09-16 12:15:09 +1000 Matthew Waters <matthew@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/isomp4/gstqtmux-doc.c:
+ * gst/isomp4/gstqtmux-doc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: make documentation happy
+ introduce a base qtmux class that we can install documentation snippets
+ on instead of duplicating across alll the isomp4 elements
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-05-28 19:40:24 +1000 Matthew Waters <matthew@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ * gst/isomp4/gstqtmuxmap.c:
+ * tests/check/elements/qtmux.c:
+ isomp4/mux: add a fragment mode for initial moov with data
+ Used by some proprietary software for their fragmented files.
+ Adds some support for multi-stream fragmented files
+ Flow is as follows.
+ 1. The first 'fragment' is written as a self-contained fragmented
+ mdat+moov complete with an edit list and durations, tags, etc.
+ 2. Subsequent fragments are written with a mdat+moof and each stream is
+ interleaved as data arrives (currently ignoring the interleave-*
+ properties). data-offsets in both the traf and the trun ensure
+ data is read from the correct place on demuxing. Data/chunk offsets
+ are also kept for writing out the final moov.
+ 3. On finalisation, the initial moov is invalidated to a hoov and the
+ size of the first mdat is extended to cover the entire file contents.
+ Then a moov is written as regularly would in moov-at-end mode (the
+ default).
+ This results in a file that is playable throughout while leaving a
+ finalised file on completion for players that do not understand
+ fragmented mp4.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:37:56 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: increase some logging on streams and sample parsing
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:35:45 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: bail out when encountering an atom with a size of 0
+ A size 0 atom means the atom extends to the end of the file. No further
+ valid atoms will ever follow. Avoids a subsequent scan for an atom from
+ one byte earlier after encountering a size 0 atom.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:33:04 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix subsequent moof parsing after moov with valid samples
+ reset the moof_offset back to its original value like is done in the
+ error case just before.
+ Fixes subsequent parsing of a moof following a moov that contains valid
+ samples in a non-streaming fragmented mp4.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:30:28 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: extend edit list when fragmented
+ When we are fragmented, the edit list may only refer to the portion of
+ the media that is in the moov. Extend the edit list stop time when we
+ if there is only one qt segment and we are reading a fragmented file.
+ Fixes playback of some fragmented mp4 files generated by proprietary
+ programs.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-09-15 14:22:13 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * meson_options.txt:
+ meson: Allow overriding qt5 feature
+ This will allow controlling that feature from gst-build
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/729>
+
+2015-11-17 19:14:01 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Implement segment query
+ Fixes #239
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/713>
+
+2020-09-14 10:15:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ rtpmp4gdepay: Allow lower-case "aac-hbr" instead of correct "AAC-hbr"
+ Various live555 based products are using the wrong "mode" string or
+ seem to assume case-insensitive matching, which is wrong.
+ Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/727>
+
+2020-05-02 02:21:00 +0200 Stefan Brüns <stefan.bruens@rwth-aachen.de>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add support for AAX encrypted audio streams
+ This is modelled after the DASH Common Encryption scheme, but is somewhat
+ simpler as more parts are fixed, i.e. just one encryption scheme.
+ The output caps are fixed to 'application/x-aavd'. All information
+ required for decryption are part of the 'adrm' atom, which is passed
+ on as a property. The property is attached to the buffer.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
+
+2020-05-02 02:20:44 +0200 Stefan Brüns <stefan.bruens@rwth-aachen.de>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Add 'aavd' and related fourcc codes for AAX encrypted audio
+ The 'aavd' box is contained in the 'stsd' sample description. The 'aavd'
+ box follows the layout of an 'mp4a' entry, i.e. it contains a single
+ standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd'
+ extension boxes.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
+
+2014-06-23 08:46:37 +0200 Haakon Sporsheim <haakon@pexip.com>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvpxdec.c:
+ * ext/vpx/gstvpxdec.h:
+ vpxdec: request a sync point on decoder errors
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/725>
+
+2020-09-13 18:31:57 +0200 Camilo Celis Guzman <camilo@pexip.com>
+
+ * gst/rtp/gstrtpvrawpay.c:
+ rtp/vrawpay: use alloc_output_buffer from base class
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/726>
+
+2020-09-07 23:20:58 +0800 Ricky Tang <ricky@deepsentinel.com>
+
+ * docs/gst_plugins_cache.json:
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix push-backchannel-buffer parameter mismatch
+ When using python, signal parameter must match with function.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/724>
+
+2020-09-10 11:24:32 +0200 Jérôme Laheurte <jlaheurte@quividi.net>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: check buffer size before dereferencing. Fixes #541
+ Some cameras (Panacast) have buggy drivers/firmware which send
+ invalid JPEG frames, containing no data, which makes jpegdec
+ crash because it assumes the frame is at least 2 bytes long.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/723>
+
+2020-09-10 11:11:00 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Improve logging of gst_flv_mux_buffer_to_tag_internal
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
+
+2020-09-09 15:12:53 +0200 Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
+ Besides looking like the correct place to put this, it allows us to drop
+ the entire aggregator queue. The old implementation only dropped at most
+ one buffer for each call of aggregate.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
+
+2020-09-08 17:35:50 +0200 Havard Graff <havard@pexip.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: plug memory-leak
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/718>
+
+2020-08-28 18:09:15 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ * ext/vpx/meson.build:
+ vp9enc: expose row-mt property
+ With recent libvpx versions, multithreading can be enabled on
+ a per-tile basis, instead of on a per tile-column basis.
+ In combination with the new tile-rows property, this allows the
+ encoder to make much better use of the available CPU power.
+ Bump minimum libvpx version to 1.7.0
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-08-28 17:45:48 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/vpx/gstvpxenc.c:
+ vpxenc: change default for deadline to good quality
+ Having the deadline set to best quality causes the encoder
+ to be absurdly slow, most real-life users will want the good
+ quality tradeoff instead.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-08-28 17:39:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp9enc: expose tile-columns and tile-rows properties
+ Based on patch by Stian Selnes <stian@pexip.com>.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-08-28 17:35:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * ext/vpx/gstvpxenc.c:
+ * ext/vpx/gstvpxenc.h:
+ vpxenc: add configure_encoder virtual method
+ For subclasses to expose format-specific properties
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-09-08 20:57:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: fix sink pad release while PLAYING
+ - Release the split mux lock while removing the probes
+ - Flush the sinkpad to unblock other pads
+ - Turn check_completed_gop into a do while statement, when
+ waking up we want to recheck whether the current GOP is
+ ready for sending
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/719>
+
+2017-10-31 09:40:33 +0000 John-Mark Bell <jmb@pexip.com>
+
+ * tests/check/elements/vp8enc.c:
+ vp8enc: improve unit tests
+ - make test_encode_simple cope with libvpx built with
+ CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at
+ runtime beyond trying to set lag-in-frames to >0, pushing a
+ buffer and catching the GST_FLOW_NOT_NEGOTIATED return.
+ - fix bitrot in test_encode_simple_when_bitrate_set_to_zero.
+ - port test_encode_simple to GstHarness and introduce a separate
+ test for the lag-in-frames property.
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/708>
+
+2020-08-21 16:03:09 +0200 Jakub Adam <jakub.adam@collabora.com>
+
+ * docs/gst_plugins_cache.json:
+ docs: Update plugin cache
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-03-24 19:35:07 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxenc.c:
+ vpx: Support GST_VIDEO_FORMAT_I422_10LE
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-03-24 17:16:59 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxenc.c:
+ vpx: Support GST_VIDEO_FORMAT_I420_10LE
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-03-23 21:44:30 +0100 Jakub Adam <jakub.adam@collabora.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvpxenc.c:
+ vp9enc: support GST_VIDEO_FORMAT_Y444
+ Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-09-08 17:30:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitlab-ci.yml:
+ ci: include template from gst-ci master branch again
+
+2020-09-08 16:58:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
=== release 1.18.0 ===
2020-09-08 00:05:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+ * .gitlab-ci.yml:
* ChangeLog:
* NEWS:
* README:
* RELEASE:
+ * docs/gst_plugins_cache.json:
* gst-plugins-good.doap:
* meson.build:
Release 1.18.0
projects / platform / upstream / gstreamer.git / blobdiff