=== release 1.22.0 ===

2023-01-23 19:29:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.22.0

2023-01-23 16:27:56 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/hr.po:
	* po/ro.po:
	* po/zh_CN.po:
	  gst-plugins-good: update translations
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3773>

2023-01-17 17:51:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-mux.c:
	* gst/matroska/webm-mux.c:
	  matroska: Add `stream-format = (string) obu-stream` to AV1 caps
	  Anything else is not allowed in Matroska/WebM.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>

2023-01-17 17:50:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/isomp4/gstqtmuxmap.c:
	* gst/isomp4/qtdemux.c:
	  isomp4: Add `stream-format = (string) obu-stream` to AV1 caps
	  Anything else is not allowed in MP4.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>

2023-01-17 15:04:10 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Add basic support for AVC-Intra video
	  AVC-Intra is a range of H.264-compliant intra-only codecs from
	  Panasonic. The codes and descriptions have been taken from VLC.
	  The (encumbered) sample I have here produces byte-stream H.264,
	  including SPS and PPS and no `avcC` box.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3739>

=== release 1.21.90 ===

2023-01-13 19:08:48 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.21.90

2023-01-12 14:32:30 -0500  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/rtp/gstrtpopuspay.c:
	  rtopuspay: Use GstStaticCaps to cache parsed caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

2023-01-12 14:25:52 -0500  Olivier Crête <olivier.crete@ocrete.ca>

	* gst/rtp/gstrtpopuspay.c:
	* tests/check/elements/rtpopus.c:
	  rtopuspay: Ignore the stereo parameter in multiopus caps
	  Also add unit tests for the various variants
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

2023-01-03 18:55:24 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: Leave original caps as-is
	  This should make it work if someone specifies stereo with MULTIOPUS
	  somehow.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

2023-01-03 18:06:56 -0500  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtpopuspay.c:
	  rtpopuspay: Return upstream channel filter based on OPUS vs MULTICAPS
	  Only allow 1 or 2 channels if the caps are OPUS, or 3+ if they are
	  MULTIOPUS.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

2023-01-03 17:44:37 -0500  Olivier Crête <olivier.crete@collabora.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  rtpopus: Put MULTIOPUS in all caps
	  The RTP payload encoding-name are always in caps in GStreamer.
	  In SDP, they are not case-sensitive, but since caps are, we need to pick
	  a caps, and we picked upper-case along time ago.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>

2023-01-11 15:41:09 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/LINGUAS:
	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/ka.po:
	* po/ky.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  gst-plugins-good: update translations
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

2023-01-11 14:53:39 +0000  Tim-Philipp Müller <tim@centricular.com>

	* scripts/dist-translations.py:
	* scripts/meson.build:
	  Fix translation pot files when creating dist tarballs
	  Add version as per Translation Project requirements and
	  also add a .pot file without the ABI suffix.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3711>

2023-01-07 20:03:31 +0100  Marek Vasut <marex@denx.de>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Disable libjpeg-turbo SIMD acceleration support for now
	  The libjpeg-turbo SIMD acceleration support suffers from multiple
	  unresolved cornercases. Disable the libjpeg-turbo for now until
	  those cornercases are resolved.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3694>

2022-08-07 14:23:04 +1000  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/hls/gsthlsdemux.c:
	  hlsdemux: Consider starting stream time in presentation offset
	  When calculating the presentation offset for CMAF input in live
	  playback, subtract the stream_time of the fragment from the
	  calculated presentation offset, so that the first fragment
	  is played at running time zero.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3680>

2022-12-22 21:23:39 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/qt/meson.build:
	* ext/qt6/meson.build:
	* meson_options.txt:
	  meson: Add an option to select the method for finding Qt
	  This is needed by Cerbero, since we want to force the use of qmake to
	  find Qt on non-Linux platforms.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3628>

2022-12-26 23:00:18 +0900  Seungha Yang <seungha@centricular.com>

	* ext/gtk/gstgtkbasesink.c:
	  gtkbasesink: Fix widget leak
	  gst_gtk_base_sink_get_widget() will increase refcount and it should
	  be released after use
	  Fixing regression introduced by the commit
	  941c0e81ddf5c0ad6a9fac59b1d6bf7723025434
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3644>

2022-12-27 23:33:31 +0900  Seungha Yang <seungha@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix string leak
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>

2022-12-27 23:26:48 +0900  Seungha Yang <seungha@centricular.com>

	* gst/rtpmanager/rtptimerqueue.c:
	  rtptimerqueue: Fix memory leak
	  Should chain up to parent's finalize
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>

2022-12-22 11:16:26 +0100  Patricia Muscalu <patricia@axis.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix buffer leak in fragment_buffers
	  When pushing buffers from one of the sink pads fail,
	  make sure that all buffers added to fragment_buffers on other pads
	  are freed as well.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3624>

2022-09-19 21:11:39 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: For video with N/1001 framerates use N as timescale instead of centiframes
	  This is recommended by various specifications for such framerates, while
	  for integer framerates we continue using centiframes to allow for some
	  more accuracy.
	  Using N means that no rounding error accumulates, eventually leading to
	  outputting a packet with a different duration.
	  Some tools such as MediaInfo determine that a stream is variable
	  framerate if any packet has a different duration than the others, and
	  there is no reason I can see for not using the full 4 bytes of
	  resolution that the mp4 timescale offers.
	  Example problematic pipeline:
	  ```
	  videotestsrc num-buffers=5001 ! video/x-raw,framerate=60000/1001,width=320,height=240 ! \
	  videoconvert ! x264enc bitrate=80000 speed-preset=1 tune=zerolatency ! h264parse ! \
	  video/x-h264,profile=high-10 ! mp4mux ! filesink location="result2.mp4"
	  ```
	  This results in a media file that MediaInfo detects as variable
	  framerate because the 5000th packet has duration 99 instead of 100.
	  With this patch, the timescale is 60000 and all packets have duration
	  1001.
	  Related issue for context: https://bugzilla.gnome.org/show_bug.cgi?id=769041
	  Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3049>

2022-12-21 02:37:58 +1100  Jan Schmidt <jan@centricular.com>

	* ext/qt/qtwindow.cc:
	  qmlglsrc: Handle HiDPI scaling
	  When calculating the capture framebuffer size, include
	  any device scaling applied to the rendered framebuffer
	  Fixes only capturing part of the window when there is
	  a global scale factor.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3612>

2022-12-21 02:32:41 +1100  Jan Schmidt <jan@centricular.com>

	* ext/qt/qtwindow.cc:
	  qmlglsrc: Unmap buffer before adding sync meta
	  Adding a sync meta to a GstBuffer requires that it
	  be writable. Mapping the buffer with the video frame API
	  holds an extra ref on the buffer, so unmap before
	  trying to modify it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3612>

2022-12-21 02:31:05 +1100  Jan Schmidt <jan@centricular.com>

	* ext/qt/gstqtsrc.cc:
	  qmlglsrc: Stop when basesrc calls unlock()
	  Instead of stopping capture when the state changes,
	  handle other cases of basesrc stopping capture by - such
	  as handling an EOS event - by implementing an unlock()
	  method
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3612>

2022-12-16 17:27:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Always use `tfdt` if available in BYTE segments
	  This reverts the decision from
	  https://bugzilla.gnome.org/show_bug.cgi?id=754230
	  where it was decided that we rather play safe and only use the `tfdt` if
	  it is "significantly different" to the sum of sample durations.
	  As the specification says
	  If the time expressed in the track fragment decode time (‘tfdt’) box
	  exceeds the sum of the durations of the samples in the preceding
	  movie and movie fragments, then the duration of the last sample
	  preceding this track fragment is extended such that the sum now
	  equals the time given in this box.
	  we have to use the `tfdt` in general to allow for it to signal gaps in
	  the stream.
	  A muxer producing fragments might not yet know the full duration of the
	  last sample of a previous fragment if the next fragment starts with a
	  gap, and knowing the actual start of the next fragment would potentially
	  require to violate latency requirements.
	  Additionally, the existence of `tfdt` allows to avoid accumulating
	  rounding errors from summing up the durations.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3586>

2022-12-16 01:00:46 -0600  A. Wilcox <AWilcox@Wilcox-Tech.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  tests: Cast drop-messages-interval type properly
	  The rtpjitterbuffer test drop_messages_interval uses a GstClockTime for
	  the message drop interval.  This property is defined as a guint.  On
	  systems with 64-bit time_t but 32-bit uint, this can cause the
	  g_object_set function to fail to read the arguments properly.
	  Fixes: #1656
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3580>

2022-12-13 12:23:56 +0100  Thibault Saunier <tsaunier@igalia.com>

	* ext/qt/qtitem.cc:
	* ext/qt6/qt6glitem.cc:
	  base:navigation: Cleanup navigation key modifiers enum
	  We were exposing the 'ALT' modifier as if we were guaranteeing its
	  accuracy but truth is we were only exposing configuration dependent
	  values.
	  Make the API simpler for now, the same way as Gtk3 was exposing it, and
	  when we have time to guarantee more values by making them take backends
	  configuration into account, we will expose those values in a accurate
	  way.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1402
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3565>

2022-12-09 11:35:25 +0100  Xabier Rodriguez Calvar <calvaris@igalia.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Clear protection events when we get new ones
	  If we keep the old events they can be end up being passed to the app, that could
	  discard the protection information because it has been seen before.
	  Drive by improvement: use g_queue_clear_full instead of foreach+clear for
	  protection events.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3547>

2022-05-22 10:46:12 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Enable packetized if sink caps contains parsed as true.
	  jpegdec is capable to parse input frames, but if jpegparse is before,
	  there's no need to reparse frames. This patch configure jpegdec as
	  packetized, skipping parsing, if negotiated sink caps has the boolean
	  field 'parsed' as true.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2464>

2022-09-27 09:44:51 +0000  Henry Hoegelow <henry@hoegelow.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Fix occasional period of silence on resume
	  According to comment in gst_pulsering_stream_latency_cb, latency updates
	  happen every 100 ms. The code in gst_pulsering_stream_latency_cb does
	  not care about the actual state of the ringbuffer, pbuf->acquired or
	  not.
	  Thus, every 100 ms the ringbuf->segdone may be set, even though the
	  object itself might be in 'destroyed' state. On next
	  gst_pulseringbuffer_acquire the segdone is not touched, so playback may
	  resume with invalid/wrong segdone value. This finally leads to a period
	  of silence playing after resuming the pipeline.
	  The problem was found on 'not-yet-released'-hardware and so far was not
	  reproducible on desktop computer.
	  Removing the callback as long as the ringbuffer is not in 'acquired'
	  state solves the problem reliably on the hardware device that the issue
	  was detected on.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3082>

2021-05-14 23:06:43 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpvp9depay.c:
	* gst/rtp/gstrtpvp9depay.h:
	  rtpvp9depay: expose keyframe-related properties
	  This simply brings in the wait-for-keyframe and request-keyframe
	  properties from rtpvp8depay.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/909>

2022-12-07 09:47:49 +0100  Jacek Skiba <jacek.skiba@consult.red>

	* gst/isomp4/qtdemux.c:
	  qtdemux: exit when protection caps are not defined during PIFF parsing
	  Reproduction testcase (uses PlayReady):
	  https://developers.canal-plus.com/rx-player/upc/?appTileLocation=[object%20Object]
	  In test streams we are using PIFF box, but caps did not had
	  present GST_PROTECTION_SYSTEM_ID_CAPS_FIELD. In consequence, invalid
	  system_id was returned which caused SIGSEGV crash.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3535>

2022-12-05 10:55:57 +0100  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux.c:
	  adaptivedemux2: Don't allow stream selection while switching periods
	  The stream selection is done on the currently outputting tracks, but in order to
	  (de)activate the backing streams we can only do it if the input and output
	  period are identical.
	  Fixes crash when doing stream selection during period migration
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3525>

2022-12-05 02:29:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development

=== release 1.21.3 ===

2022-12-05 01:28:21 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.21.3

2022-12-04 12:25:41 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	  Remove ChangeLog files from git repository
	  This information is tracked fully in the git repository, so
	  no point having the ChangeLog duplicate it, and it interferes
	  with grepping the repository.
	  We are going to create the ChangeLogs on the fly when generating
	  tarballs going forward (with a limited history), since it's still
	  valuable for tarball consumers to be able to easily see a list of
	  recent changes.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/73
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3521>

2022-11-07 00:10:39 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* scripts/gen-changelog.py:
	  meson: Generate ChangeLog files for release tarballs on dist
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3521>

2022-12-04 11:44:17 +0000  Philippe Normand <philn@igalia.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: Fix handling of headers advertising 32bps
	  According to the flac bitstream format specification, the sample size in bits
	  corresponding to `111` is 32 bits per sample.
	  https://xiph.org/flac/format.html#frame_header
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3517>

2022-12-02 12:15:34 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2src.c:
	  v4l2src: Fix crash in renegotiation
	  This regression was introduce by fix for making buffer pool thread safe. When
	  we renegotiate, the pool will be setup after we set the format. But the code
	  has been simplified to only get the pool once before, which caused a null
	  pointer deref.
	  Fixes 94ba019 ("v4l2: Fix SIGSEGV on 'change state' during 'format change'")
	  Related to !3481
	  Fixes #1626
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3513>

2022-12-01 10:34:10 +0100  Aleksandr Slobodeniuk <aslobodeniuk@fluendo.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: fix seek event leaks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3500>

2022-11-30 14:32:52 +0100  Bo Elmgreen <bo.elmgreen@gmail.com>

	* ext/qt/gstqtglutility.cc:
	* ext/qt6/gstqt6glutility.cc:
	  qt: deactivate context if fill_info fails
	  Now the OpenGL context is deactivated if call to gst_gl_context_fill_info()
	  fails in gst_qt_get_gl_wrapcontext(), preventing that the context is left
	  activated, which could lead to invalid memory reads.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3492>

2022-11-28 22:26:50 +0000  Pawel Stawicki <stawel+gstreamer@gmail.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2transform.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: Fix SIGSEGV on 'change state' during 'format change'
	  Ensure all access to v4l2object->pool imply taking a lock and a hard ref on the pool
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3481>

2022-11-22 11:32:57 -0500  Matt Crane <matt@standard.ai>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Support disabling late adjustment of ntp-64 header ext
	  Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
	  extension timestamp is updated with the actual NTP time before sending
	  the packet. However, there are some circumstances where we would like
	  to preserve the original timestamp obtained from reference timestamp
	  buffer metadata.
	  This commit provides the ability to configure whether or not to update
	  the ntp-64 header extension timestamp with the actual NTP time via the
	  update-ntp64-header-ext boolean property. The property is also exposed
	  via rtpbin. Default property value of TRUE will preserve existing
	  behavior (update ntp-64 header ext with actual NTP time).
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>

2022-10-06 17:08:54 +1100  Matthew Waters <matthew@centricular.com>

	* ext/meson.build:
	* ext/qt6/gstplugin.cc:
	* ext/qt6/gstqml6glsink.cc:
	* ext/qt6/gstqml6glsink.h:
	* ext/qt6/gstqsg6glnode.cc:
	* ext/qt6/gstqsg6glnode.h:
	* ext/qt6/gstqt6element.cc:
	* ext/qt6/gstqt6elements.h:
	* ext/qt6/gstqt6gl.h:
	* ext/qt6/gstqt6glutility.cc:
	* ext/qt6/gstqt6glutility.h:
	* ext/qt6/meson.build:
	* ext/qt6/qt6glitem.cc:
	* ext/qt6/qt6glitem.h:
	* meson_options.txt:
	* tests/examples/meson.build:
	* tests/examples/qt6/meson.build:
	* tests/examples/qt6/qmlsink/main.cpp:
	* tests/examples/qt6/qmlsink/main.qml:
	* tests/examples/qt6/qmlsink/meson.build:
	* tests/examples/qt6/qmlsink/play.pro:
	* tests/examples/qt6/qmlsink/qmlsink.qrc:
	  add new plugin for Qt 6 rendering inside a QML scene
	  - Based heavily on the existing Qt5 integration however:
	  - The sharing of OpenGL resources is slightly different
	  - The integration with the scengraph is a bit different
	  - Wayland, XCB and KMS have been smoke tested.  Android, MacOS/iOS,
	  Windows may or may not work.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3281>

2022-11-21 15:35:58 +0800  Elliot Chen <elliot.chen@nxp.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: bypass check some transfer types in colorimetry
	  v4l2 will report fail for some streams whose colorimetry value such as 2:4:8:3.
	  Can bypass check these transfer types in colorimetry to avoid error.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3440>

2022-10-27 17:30:28 +0200  Johan Sternerup <johast@axis.com>

	* gst/rtpmanager/rtpsession.c:
	* tests/check/elements/rtpsession.c:
	  Use the correct SSRC(s) when routing a RTCB FB FIR
	  Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
	  using the "media source" component of the RTCP FB message. However,
	  according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
	  to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
	  a specific GstForceKeyUnit event is sent for every ssrc.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>

2022-11-23 16:37:40 +1100  Jan Schmidt <jan@centricular.com>

	* gst/rtpmanager/rtpsource.c:
	  rtpsource: Track the seqnum for senders
	  RTP source statistics are tracked for local senders by
	  treating them as a receiver of their own outbound packets.
	  Accordingly, track the highest packet seqnum so that the
	  packets-lost calculation generates a sensible number instead
	  of always reporting -$number_of_packets_sent
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>

2022-11-22 21:57:16 +1100  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/meson.build:
	  adaptivedemux2: Add GStreamer to the deps list
	  Explicitly dep on GStreamer so as not to accidentally
	  link to the system version in a git build
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3453>

2022-11-15 11:56:35 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't replace 404 errors with "no auth protocol found"
	  When getting a "404 Not Found" response from the DESCRIBE request, the
	  source produced a "No supported authentication protocol was found" error
	  instead of passing on the 404, which was confusing.
	  Only produce this error message when we're handling a response of "401
	  Unauthorized" without a compatible WWW-Authenticate header.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>

2022-11-21 17:24:38 +0100  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux-stream.c:
	  adaptivedemux2: Don't leak caps in debug statements
	  Instead just directly use the stream object (which will report the caps)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

2022-11-21 17:23:44 +0100  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux-stream.c:
	  adaptivedemux2: Don't leak tags
	  If we got them from GstStream, we should unref them when done
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

2022-11-21 17:23:10 +0100  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux-stream.c:
	* ext/adaptivedemux2/gstadaptivedemux.c:
	  adaptivedemux: Use gst_clear_tag_list_where applicable
	  Clearer and ensures fields are reset
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

2022-11-21 15:11:21 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't leak sticky events
	  We have incremented the reference 2 lines above, and
	  gst_pad_store_sticky_event() does not take a reference, therefore release it
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>

2022-11-15 21:42:34 +1100  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemuxutils.c:
	  adaptivedemux2: Fix sticky event storage.
	  Use the new gst_event_type_to_sticky_ordering() method to retrieve
	  the order that sticky events should be sent / stored in, instead
	  of assuming it's the event type value.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3387>

2022-11-15 13:50:13 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2object.c:
	  video: Add arbitrary tile dimensions support
	  In current tile representation, only tiles with power of two
	  width and height in bytes are supported. This limitation
	  prevents adding more complex tiles formats.
	  In this patch, we deprecate tile_ws and tile_hs from GstVideoFormatInfo and
	  replace if with an array of GstVideoTileInfo. Each plane tiles are then
	  described with their pixels width/height, line stride and total size.
	  The helper gst_video_format_info_get_tile_sizes() that depends on the
	  deprecated API is also being removed. This can simply be removed as it wasn't
	  in any stable release yet.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>

2022-09-09 13:13:32 +0300  Vivia Nikolaidou <vivia@ahiru.eu>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Avoid assertion when WAITING_GOP_COLLECT on reference context
	  I have seen a backtrace out in the wild where this happened. Maybe after
	  receiving EOS and stream-start on the reference context.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3005>

2022-11-14 13:53:02 +0100  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstimagesequencesrc.c:
	  imagesequencesrc: Don't leak caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3428>

2022-05-03 18:36:54 +0200  Enrique Ocaña González <eocanha@igalia.com>

	* ext/adaptivedemux2/hls/gsthlsdemux.c:
	* ext/adaptivedemux2/hls/gsthlsdemux.h:
	  hlsdemux2: Expose EXT-X-PROGRAM-DATE-TIME as tags.
	  This allows an application to use timestamps associated
	  with fragments.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1417>

2022-11-17 20:59:35 +1100  Matthew Waters <matthew@centricular.com>

	* gst/isomp4/atoms.c:
	  qtmux: use trun with multiple entries in more cases
	  The only case where we definitely need to write a new trun is when the
	  data_offset value does not match the end of the list of entries.
	  Needing multiple trun atoms is required when interleaving multiple
	  streams together.
	  All other cases can be covered by adding more entries to the existing
	  trun atom.
	  Fixes playback of fragemented mp4 in ffplay and chrome.
	  Using e.g. mp4mux fragment-duration=1000 fragment-mode=dash-or-mss
	  and
	  mp4mux fragment-duration=1000 fragment-mode=first-moov-then-finalise
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3426>

2022-10-26 07:04:32 +0200  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux.c:
	  adaptivedemux2: Notify that we are streams-aware
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>

2022-11-16 13:33:39 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix regression when using hostname in the location property
	  When the address can't be parsed as an IP address, it should just be
	  treated as a hostname and used as-is.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1576
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3420>

2022-11-13 19:19:41 -0500  F. Duncanh <fduncanh@gmail.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: Fixes three H.264/HEVC ITU conformance tests
	  Postpone the cleanup of any consecutive sequence of lost frames
	  which starts at frame 0, until frame 100 is dequeued from driver.
	  This allows fluster tests JVT/CVWP2_TOSHIBA_E, JVC/CVWP3_TOSHIBA_E
	  and HEVC/POC_A_Bossen_3 that sends out-of-order frames to successfully
	  complete  (e.g., test of Amphion vpu driver).
	  Fixes #1569
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3398>

2022-11-15 17:26:18 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/rtpjitterbuffer.c:
	  rtpjitterbuffer: Add test for rescheduling timers to negative times
	  This tests the changes introduced by 4d3b8d1129d8b863e4156cd0334e93257b9d0cc4
	  for issue #571.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3416>

2022-09-07 11:07:40 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Reschedule timers when updating their offset
	  As EXPECTED timers are skipped the order of the timers relative to each
	  other can change if there are EXPECTED timers and rescheduling needs to
	  happen.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1422
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3416>

2022-11-15 21:22:17 +0530  Sanchayan Maity <sanchayan@asymptotic.io>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Do not run all typefinders for all output
	  In order to figure out if the "raw" audio contained within the wav
	  container is actually DTS, wavparse calls the typefinder helper
	  except that means it runs all typefinders.
	  Since it only cares about checking for DTS, we should only run the
	  audio/x-dts typefinder (if present). Commit 858e516 did not really
	  fix things.
	  Use the new type helper with the caps to fix this.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3417>

2022-11-14 21:19:12 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Consistently set seqnums on events
	  Set udpsrc seqnums on all events sent to the udpsrc's, and before
	  forwarding events out of rtspsrc set the latest seek seqnum on them if
	  any.
	  Also produce a consistent seqnum in rtspsrc from the very beginning.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>

2022-11-14 19:14:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Make segment event writable before overriding the seqnum and use the proper API to do so
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>

2022-11-14 19:10:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Intercept and handle events when using no manager too
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>

2022-11-14 19:08:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't blindly copy over sticky events from manager pad to external source pad
	  This would get around the code that modifies some events when they go
	  through the ghost pad's proxypad. Instead go via the event function.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>

2022-11-14 19:03:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't make udpsrc segment events writable just to retrieve their seqnum
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>

2022-11-14 18:40:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Reset EOS flag also on FLUSH_STOP and not only on ssrc-active
	  Also don't bother not sending EOS if EOS was sent already:
	  gst_pad_push_event() takes care of that for us already.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>

2022-11-15 13:38:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: fix docs - changing images at runtime is supported
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3415>

2022-11-13 08:38:25 +0100  Edward Hervey <edward@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Unlock timer waits on flushing
	  If there is a pending EOS wait for example, we would never unblock on flushing
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3401>

2021-10-14 11:56:58 +0200  Rafał Dzięgiel <rafostar.github@gmail.com>

	* tests/check/elements/dash_mpd.c:
	  tests: Add DASH MPD baseURL with query test
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>

2021-10-14 10:12:51 +0200  Rafał Dzięgiel <rafostar.github@gmail.com>

	* ext/adaptivedemux2/dash/gstmpdparser.c:
	  mpdparser: Fix missing baseURL query
	  When no initializationURL or mediaURL, return baseURL that also
	  contains original URI query if available. This fixes a problem
	  where URI query was being omitted in the HTTP requests.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>

2021-10-14 10:09:31 +0200  Rafał Dzięgiel <rafostar.github@gmail.com>

	* ext/adaptivedemux2/dash/gstmpdclient.c:
	* ext/adaptivedemux2/dash/gstmpdparser.c:
	* ext/adaptivedemux2/dash/gstmpdparser.h:
	  mpdparser: Be consistent about returning duplicated URL
	  Instead of returning a "const gchar" or a "gchar" that should not be freed, always
	  return a duplicated string as those functions were used together with g_strdup anyway.
	  This is needed to prepare support for returning modified strings in next commit.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>

2021-10-14 10:18:40 +0200  Rafał Dzięgiel <rafostar.github@gmail.com>

	* ext/adaptivedemux2/dash/gstmpdparser.c:
	  mpdparser: Return correct mediaURL value
	  This fixes a problem where get_mediaURL was returning NULL when segmentURL
	  was unavailable instead of baseURL as a fallback.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>

2022-11-14 12:51:19 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Handle element's duration query.
	  This is small regression from commit f7abd81a.
	  When calling `gst_element_query()` no pad is associated with that query, but the
	  current code always forwards the query to the associated pad, which is NULL in
	  previous case. This patch checks for the pad before forwarding the query.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3404>

2022-11-13 12:18:54 +0100  Rafał Dzięgiel <rafostar.github@gmail.com>

	* ext/adaptivedemux2/gstadaptivedemux.c:
	  adaptivedemux2: Fix parent object leak
	  gst_object_get_parent() method is transfer-full, thus unref is needed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3396>

2022-10-10 19:58:12 +0100  Colin Kinloch <colin.kinloch@collabora.com>

	* gst/videobox/gstvideobox.c:
	* gst/videocrop/gstvideocrop.c:
	  videocrop, videobox: Simplify navigation event handling and support touch events
	  Signed-off-by: Colin Kinloch <colin.kinloch@collabora.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3053>

2022-09-19 18:06:44 +0100  Colin Kinloch <colin.kinloch@collabora.com>

	* gst/videofilter/gstvideoflip.c:
	  videoflip: Use gst_video_orientation_from_tag to parse orientation
	  Signed-off-by: Colin Kinloch <colin.kinloch@collabora.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3053>

2022-11-09 12:04:08 +0000  Christian Wick <c.wick@mail.de>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	* tests/examples/rtsp/test-onvif.c:
	  rtspsrc: Introduce new action signal `push-backchannel-sample` with correct ownership semantics
	  Signals are not supposed to take ownership of their arguments but only
	  borrow them for the scope of the signal emission.
	  The old action signal `push-backchannel-buffer` is now marked deprecated.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3363>

2022-11-08 02:08:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* meson.build:
	  Back to development
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3358>

=== release 1.21.2 ===

2022-11-07 23:53:59 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* docs/gst_plugins_cache.json:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.21.2

2022-11-07 23:53:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	  Update ChangeLogs for 1.21.2

2020-09-02 10:49:40 +0100  Justin Chadwell <me@jedevc.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: use unsigned int types to store result of QT_UINT32
	  In a few cases throughout qtdemux, the results of QT_UINT32 were being
	  stored in a signed integer, which could cause subtle bugs in the case of
	  an integer overflow, even allowing the the result to equal a negative
	  number!
	  This patch prevents this by simply storing the results of this function
	  call properly in an unsigned integer type. Additionally, we fix up the
	  length checking with stsd parsing to prevent cases of child atoms
	  exceeding their parent atom sizes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3344>

2022-11-04 17:48:01 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ext/qt/gstqtglutility.cc:
	  qt: initialize GError properly in gst_qt_get_gl_wrapcontext()
	  Spotted by Claus Stovgaard.
	  Fixes #1545
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3327>

2022-11-04 11:10:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Add durations to raw audio buffers from the raw audio adapter in prefill mode
	  This ensures that a duration can also be calculated and stored for the
	  last buffer at EOS.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3321>

2022-11-04 10:49:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Release object lock before posting an error message
	  GST_ELEMENT_ERROR() also takes the object lock and this would then
	  deadlock.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3321>

2022-11-03 14:08:57 +0100  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstimagesequencesrc.c:
	  imagesequencesrc; Fix leaks
	  * The path was leaked
	  * The custom buffer was never freed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>

2022-11-03 14:08:02 +0100  Edward Hervey <edward@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix cenc-related leaks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>

2022-11-03 14:06:58 +0100  Edward Hervey <edward@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Don't leak metas
	  There is no correlation between the frame being NULL and the metas not being
	  present.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>

2022-10-31 16:08:23 +0100  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux-period.c:
	* ext/adaptivedemux2/gstadaptivedemux.c:
	  adaptivedemux2: Fix collection leaks
	  * The collection on the period was never unreffed
	  * The collection in the message handler was never unreffed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>

2022-11-05 03:23:43 +1100  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/mss/gstmssdemux.c:
	  mssdemux2: Update for adaptivedemux2 refactoring
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>

2022-11-03 01:48:08 +1100  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/dash/gstdashdemux.c:
	* ext/adaptivedemux2/gstadaptivedemux-private.h:
	* ext/adaptivedemux2/gstadaptivedemux-stream.c:
	* ext/adaptivedemux2/gstadaptivedemux-stream.h:
	* ext/adaptivedemux2/gstadaptivedemux.c:
	* ext/adaptivedemux2/gstadaptivedemux.h:
	* ext/adaptivedemux2/hls/gsthlsdemux.c:
	  adaptivedemux2: Move stream_seek() to the Stream class
	  Move the last stream specific vfunc from the demux
	  class to the stream class.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>

2022-08-21 04:31:53 +1000  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/dash/gstdashdemux.c:
	* ext/adaptivedemux2/gstadaptivedemux-private.h:
	* ext/adaptivedemux2/gstadaptivedemux-stream.c:
