=== release 1.28.3 ===

2026-05-11 18:28:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.28.3

2026-04-16 17:03:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check for minimum stride requirements and width/height constraints with uncompressed video
	  And return an explicit error on mismatches.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5043
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11591>

2026-05-08 22:58:24 +1000  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/hls/gsthlsdemux.c:
	  hlsdemux2: Abort wait for variant playlist on shutdown
	  If the element is stopped while waiting for the variant playlist,
	  exit the loop.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11572>

2026-05-08 22:24:52 +1000  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux-stream.c:
	  adaptivedemux2: Fix mismatched clock-time vs clock-stime check
	  Adaptive demux stream start_position is an unsigned clock-time value,
	  it should not be compared against GST_CLOCK_STIME_NONE
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11572>

2026-05-08 22:09:46 +1000  Jan Schmidt <jan@centricular.com>

	* ext/adaptivedemux2/hls/gsthlsdemux-stream.c:
	* ext/adaptivedemux2/hls/gsthlsdemux.c:
	* ext/adaptivedemux2/hls/gsthlsdemux.h:
	  hlsdemux2: Error out cleanly on resync failure
	  Instead of asserting when the variant stream doesn't have a current
	  position to resynchronise to after a discontinuity, throw an element
	  error instead.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11572>

2026-04-09 09:59:52 +0900  Shigeharu Kamiya <s_kamiya@toa.co.jp>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: include user-agent property in HTTP tunnel requests
	  The user-agent property is correctly sent in regular RTSP
	  requests via gst_rtspsrc_init_request(), but was missing from
	  HTTP tunnel GET and POST requests. Add it to the connection's
	  extra HTTP headers when tunneled mode is configured.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11286>

2026-05-06 19:50:35 +0100  Arthur Chan <arthur.chan@adalogics.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: Fix memory leak when file is corrupted
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11564>

2026-04-07 13:09:24 +0100  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	  meson: Require C std gnu11 or c11
	  When using gcc or clang, gnu11 will be used, and when using MSVC c11
	  will be used which will pass /std:c11. This provides us with the
	  `restrict` keyword on all supported platforms.
	  We can do this now because we require Visual Studio 2019.
	  This feature was added in Meson 1.3
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11563>

2026-05-06 19:49:33 +0100  Arthur Chan <arthur.chan@adalogics.com>

	* gst/matroska/ebml-read.c:
	  matroska: Fix wrong object type bug
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11561>

2026-04-30 12:57:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Remove assertion about upstream file size
	  Upstream can report bogus values and we shouldn't assert on that but just ignore
	  it in that case.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5077
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11506>

2026-04-21 10:43:37 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  gstrtspsrc: mki is optional upon crypto update
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5050
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11448>

2026-04-04 20:34:19 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/osxaudio/gstosxcoreaudiocommon.h:
	  applemedia: Require Xcode 12.4 for all builds
	  Xcode 12.4 is the newest Xcode that you can install on macOS 10.15. We
	  require Xcode 12.4, but we only support macOS builds on it. For iOS,
	  tvOS, etc a newer (currently unspecified) Xcode version is required.
	  Fixes #4941
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11369>

2026-04-20 15:36:34 +0900  Jeongmin Kwak <jeongmin.kwak@lge.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: qtdemux: Add bounds checks for ESDS descriptors
	  Validate minimum sizes before accessing descriptor fields to prevent
	  out-of-bounds reads on truncated or malformed MP4 files.
	  - ES_DESCRIPTOR_TAG: require 3 bytes
	  - DECODER_CONFIG_DESC_TAG: require 13 bytes
	  - SL_CONFIG_DESC_TAG: require 1 byte
	  Skip short descriptors instead of reading invalid memory.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11382>

2025-10-15 11:16:46 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2transform.c:
	  v4l2transform: release input buffers earlier
	  The transform operation is synchronous. So the input buffer is no longer needed
	  once the output buffer is ready. Release it immediately so it can be reused by
	  the upstream pipeline.
	  This can reduce the number of buffers needed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11368>

2025-10-15 14:23:03 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2bufferpool.h:
	  v4l2: pool: add helper to release buffers
	  Factor out the code to dequeue and release buffers that are ready.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11368>

2026-04-14 15:43:58 +0200  Tobias Koenig <tobias.koenig@kdab.com>

	* ext/qt6/qt6glitem.cc:
	  qml6glsink: Fix redraw issues on buffer change
	  Mark the texture as dirty if the new buffer is different from
	  the previous one to make sure the item is rerendered.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11338>

2026-04-14 09:04:15 +0200  Michiel Westerbeek <happylinks@gmail.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: recover from invalid av_bps for PCM
	  Some PCM WAV files in the wild have an av_bps field (nAvgBytesPerSec)
	  that doesn't match blockalign * rate — only the metadata is wrong,
	  the PCM data is valid. Previously wavparse rejected these with a
	  fatal stream error.
	  For PCM, av_bps is fully determined by blockalign * rate, so
	  recompute to the authoritative value and warn. wav->bps is used at
	  :367 and :734 to translate timestamps to byte offsets for seeking,
	  so simply accepting the bogus value would silently produce wrong
	  seeks.
	  Non-PCM formats keep the strict rejection: for compressed formats
	  (ADPCM, GSM, etc.) the actual av_bps is much less than
	  blockalign * rate, so recomputing wouldn't give a correct value.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11331>

2026-04-10 17:02:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix divide by zero if VPRP contains fields==0
	  Thanks to RTV for reporting.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5027
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11305>

2026-04-03 15:35:38 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: object: Fix caps filtering in caps negotiation
	  The dmabuf caps feature was not properly added to the probed caps. This would
	  lead to caps negotiation failure when trying to force DMABuf using a caps filter
	  after the stateful v4l2h*dec elements.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11288>

2026-04-07 20:07:19 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.28.2

=== release 1.28.2 ===

2026-04-07 20:02:23 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.28.2

2026-04-01 13:39:27 +0300  Cameron O'Neal <tzusec@gmail.com>

	* gst/matroska/matroska-read-common.c:
	  matroskademux: Add missing parenthesis when calculating bz2 buffer sizes
	  This fixes a heap buffer overflow.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5009
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11263>

2026-04-01 13:44:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Use prepend+reverse instead of append when building the cues list
	  append has quadratic behaviour and this can matter if there are a lot of cues.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11262>

2026-04-01 13:42:51 +0300  Cameron O'Neal <tzusec@gmail.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix integer overflow when checking available buffer size for reading cues
	  See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5009
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11262>

2026-03-19 12:24:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: Avoid assertions on corrupted streams
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4975
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11258>

2026-03-13 11:56:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add various integer overflow and bounds checks to uncompressed video handling
	  Fixes ZDI-CAN-29392
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4963
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11252>

2026-03-19 11:00:50 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix bit pattern check for omitted audio channels map
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11253>

2026-03-19 10:46:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Avoid out-of-bounds reads and writes of 64 item audio channel positions array
	  When parsing both the chan and chnl boxes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11253>

2026-03-19 10:34:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Validate chnl defined layout before using it to index the layouts array
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4976
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11253>

2026-03-19 10:25:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Avoid division by zero if 0 audio channels are signalled
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4977
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11253>

2026-04-03 16:15:26 +0200  Frédéric Chanal <frederic.chanal@ubicast.eu>

	* gst/flv/gstflvmux.c:
	  flvmux: fix race condition on caps get and check
	  Apart from fixing a race condition this commit also removes useless code that
	  remained after !10071 which removed the use of caps for audio and video sink
	  pads.
	  The race condition appears with this sequence:
	  [thread 1] the thread in which the mux video sink element runs
	  [thread 2] the thread running the aggregator task (gst_aggregator_loop)
	  * [thread 1] GST_EVENT_CAPS is send on muxer's video pad
	  * [thread 1] eventually `gst_aggregator_default_sink_event_pre_queue()` is called
	  and the event is pushed onto internal queue for later processing
	  * [thread 1] then `SRC_BROADCAST()` is called
	  => race appears here if [thread 2] is scheduled
	  * [thread 2] `SRC_WAIT()` from `gst_aggregator_wait_and_check()` is released and
	  this function is revaluate, and return TRUE with timeout being 1 (as both audio
	  and video codec are valid, and we are ready to mux)
	  * [thread 2] The `gst_flv_mux_aggregate()` is eveantually called ->
	  `gst_flv_mux_prepare_src_caps()` -> `gst_flv_mux_create_metadata()`
	  * [thread 2] in `gst_flv_mux_create_metadata(),` the caps for the first video
	  pad (`default_video_track_pad`) are retrieved but at this point the caps are
	  NULL as the `GST_EVENT_CAPS` has not yet been stored in the sticky datastore
	  of the pad
	  * [thread 2] g_assert(caps) triggers
	  Thus, this leftover code was not merely useless—it directly caused a race
	  condition.
	  Reproduction scenario: run the following pipeline on 1.28.1
	  ```
	  taskset -c 0 gst-launch-1.0 -e videotestsrc ! "video/x-raw, format=(string)I420, \
	  width=(int)1920, height=(int)1080, framerate=(fraction)30/1" ! openh264enc ! \
	  h264parse ! flvmux streamable=true name=mux ! fakesink \
	  audiotestsrc is-live=true wave=ticks ! mux. \
	  --gst-debug=3,flvmux:6,aggregator:6
	  ```
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11235>

2026-04-02 16:22:58 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix const-correctness issue around strchr() usage
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11217>

2026-03-27 14:41:03 +1100  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Discard early data in ONVIF mode
	  We need the PLAY response from the server in order to generate the
	  segment for incoming data. Any packets that arrive before the server
	  play response can't be handled until we know that - they either need
	  to be discarded or held back somewhere and processed later.
	  Discarding is the easier thing, so I did that first
	  Fixes #4874
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11197>

2025-01-08 20:15:03 +1100  Jan Schmidt <jan@centricular.com>

	* ext/qt/qtitem.cc:
	  qt5: Avoid parsing caps on every buffer
	  Optimisation: Don't update the caps on the texture every
	  frame - only do it when they change. This avoids
	  calling gst_video_info_from_caps() for every frame
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11184>

2025-01-08 20:15:03 +1100  Jan Schmidt <jan@centricular.com>

	* ext/qt6/qt6glitem.cc:
	  qt6: Avoid parsing caps on every buffer
	  Optimisation: Don't update the caps on the texture every
	  frame - only do it when they change. This avoids calling
	  gst_video_info_from_caps() every frame.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11184>

2026-03-26 07:10:43 +1100  Jan Schmidt <jan@centricular.com>

	* ext/qt/qtitem.cc:
	  qmlglsink: Fix for caps tracking on multiple setCaps calls
	  Apply the same fix to qt5 as was put into qt6 in !10959,
	  to avoid a sequence where the caps don't update correctly
	  when incoming caps get toggled between caps-A, caps-B and back
	  to caps-A without a buffer in between.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11184>

2026-03-27 16:23:09 +0100  Per Enstedt <per.enstedt@svt.se>

	* gst/audiofx/audioinvert.c:
	  audioinvert: fix float truncation in transform_float
	  The val variable was declared as glong, causing float samples to be
	  truncated to integers before being written back to the output buffer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11171>

2026-03-24 15:17:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Error out instead of trying to handle a truncated stts box
	  Any of the handful of sample boxes could potentially be truncated and handling
	  one of them in a special way but none of the others doesn't really improve the
	  situation but complicates the special cases the sample table code needs to
	  be able to handle.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11164>

2026-03-23 19:36:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check that big enough stco/stsz are available when parsing sample tables
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11164>

2026-03-23 19:30:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Don't store 64 bit integers in 32 bit integers to avoid overflows
	  All these numbers should never become even close to 2**32 but by storing them in
	  too small integers we end up with overflows that end up at least in assertions.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4985
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11164>

2026-03-23 13:59:17 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Avoid integer overflows when handling transform matrices
	  See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4985
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11164>

2026-03-23 21:55:21 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Preserve Metas and Flags when doing row alignment
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11156>

2026-01-22 19:21:08 +0100  Alicia Boya García <aboya@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: fix handling of in-between fragments without tfdt
	  Some tools, particularly MP4Box, will generate a tfdt box only for the
	  first fragment of each segment, not for each fragment.
	  qtdemux used to be able to handle files like this, but for push-mode
	  this was accidentally broken in 794dcfbfee7cbe04a3cdbbdf7d7b7da978a7b42c.
	  That commit made it so that the sample table is cleared between
	  fragments, avoiding what otherwise would be functionally a memory leak
	  when playing long or live streams in constrained devices.
	  The existing code in qtdemux relied on the sample table to obtain the
	  DTS for the next fragment. Not persisting the sample table in push-mode
	  is still desirable so at a high level the fix is simple: keep track of
	  the running DTS separate from the sample table. This patch does that.
	  Unfortunately, the code for deciding the start DTS of a trun has gotten
	  increasingly complex and hard to analyze over the years. Instead of
	  adding another special case to it I have tried to clean it up. The new
	  code represents my understanding of what the intentions of the current
	  special cases and workarounds are, flattened and sorted by priority.
	  I have added a .mp4 and a DASH .mpd file created with MP4Box that
	  reproduces the issue fixed by this patch.
	  I do not recommend intentionally deploying on the wild files where only
	  some fragments have tfdt. Derived specifications (e.g. the MSE
	  bytestream) forbid it, it makes seeks slower and harder, and all you get
	  are meager savings of 8 bytes per fragment. This probably also explains
	  how the bug has gone unnoticed for so many years: I only created such a
	  file accidentally while trying to make a test vector for a different bug.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11067>

2026-03-18 23:43:06 +0100  Tim-Philipp Müller <tim@centricular.com>

	* po/LINGUAS:
	* po/kk.po:
	* po/zh_CN.po:
	  gst-plugins-good: update translations
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11077>

2026-03-17 09:29:30 +0100  Tobias Rapp <t.rapp@noa-archive.com>

	* gst/wavenc/gstwavenc.c:
	  wavenc: Skip writing empty LIST INFO chunk
	  When wavenc receives metadata from upstream elements but there are no
	  corresponding INFO tags defined the element did append an empty LIST
	  INFO chunk.
	  Add a follow-up check to verify whether tags have actually been written
	  to the chunk buffer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11058>

2026-03-12 14:06:42 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	* tests/check/elements/qtdemux.c:
	  qtdemux: Don't immediately push segment after moov in push mode for fmp4
	  In case of fragmented MP4 we would otherwise push a segment with potentially
	  wrong cslg shift and by that produce shifted running / stream times.
	  Simply not doing it after parsing the moov and waiting until the first buffer
	  should be output is sufficient to solve this and makes qtdemux behave the same
	  way in push and pull mode with regards to the segment and running / stream
	  times.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11055>

2026-03-16 17:32:09 +0000  Mattia <mattialeon.dm@protonmail.com>

	* ext/adaptivedemux2/gstadaptivedemux.c:
	  hlsdemux2: fix seekable range for live HLS streams
	  Fixes #4969
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11051>

2026-03-10 19:16:55 +0100  László Károlyi <laszlo@karolyi.hu>

	* gst/replaygain/gstrgvolume.c:
	* tests/check/elements/rgvolume.c:
	  rgvolume: don't apply dBSPL reference level compensation for LUFS values
	  Tools like loudgain write REPLAYGAIN_REFERENCE_LOUDNESS as a negative
	  LUFS value (e.g. -18.0), whereas traditional ReplayGain uses a positive
	  dBSPL value (e.g. 89.0). The compensation formula
	  gain += RG_REFERENCE_LEVEL - reference_level
	  would compute 89 - (-18) = +107 dB, pushing the gain out of VALID_GAIN's
	  ±60 dB range and causing it to be silently discarded, falling back to
	  0 dB and ignoring the stored gain entirely.
	  Fix this by skipping the dBSPL compensation when the stored reference
	  level is <= 0, as that unambiguously indicates LUFS format. When no
	  REPLAYGAIN_REFERENCE_LOUDNESS tag is present at all (e.g. as written by
	  rsgain), has_ref_level is FALSE and compensation is also skipped, which
	  is correct and unchanged behaviour.
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2820
	  See also: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2821
	  Assisted-By: Claude:claude-4.6-sonnet claude-code
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11043>

2026-03-12 08:49:03 +0100  Tobias Rapp <t.rapp@noa-archive.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix parsing of RF64 wave files
	  Avoid clipping the chunk size for data chunks. Fix comparing the
	  unsigned size64 with a signed constant.
	  Also allows parsing of RIFF files with data between 2-4 GiB.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2853
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11028>

2026-03-11 12:39:37 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Avoid overflow in length when setting ignore-length=true
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4957
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11026>

2026-03-12 20:50:28 +0100  Vivienne Watermeier <vwatermeier@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: fix invalid WebVTT timestamps
	  gst_qtdemux_process_buffer_wvtt() is relying on buffer timestamps to
	  write cue times into WebVTT samples.
	  However, since we now handle timestamps in qt streamtime to avoid
	  rounding, no timestamps are set on buffers at that point in time, and so
	  we write nonsense times into the samples.
	  This is fixed by just passing streamtime timestamps into 'process_func'.
	  I've checked the other functions for similar oversights; none of them
	  use buffer timestamps.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4954
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11019>

2026-03-03 16:45:03 +0700  sergey radionov <rsatom@gmail.com>

	* ext/qt6/qt6glitem.cc:
	  Qt6GLVideoItem: caps update fixed
	  on multiple Qt6GLVideoItemInterface::setCaps calls
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4944
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10968>

2026-03-02 09:59:10 +0000  James Liu <hmliu6@gmail.com>

	* gst/rtp/gstrtph264depay.c:
	  rtph264depay: fix invalid memory access in gst_rtp_h264_finish_fragmentation_unit
	  When buffer mapping fails or buffer size is smaller than sync_bytes size,
	  the process could crash due to invalid memory access.
	  Handle buffer mapping failure and validate buffer size before accessing
	  buffer data to prevent crashes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10960>

2026-01-29 12:34:46 +0100  Fabian Orccon <forccon@fluendo.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  all: GThreadFunc return type fixes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10955>

2026-02-27 17:13:27 +0000  James Liu <hmliu6@gmail.com>

	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  rtptwcc: fix feedback packet count wrapping at 255
	  The feedback packet count was incorrectly using modulo (%) with
	  G_MAXUINT8 (255), which caused it to skip the value 0xFF. According
	  to the TWCC spec (draft-holmer-rmcat-transport-wide-cc-extensions-01),
	  the feedback message sequence number must be between 0 and 255.
	  Changed to use bitwise AND (&) with G_MAXUINT8 to ensure the full
	  0-255 range is used correctly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10925>

2026-02-26 01:53:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.28.1

=== release 1.28.1 ===

2026-02-26 01:44:06 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.28.1

2026-01-12 14:26:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Avoid integer overflow and out-of-bounds read when parsing adtl chunks
	  Fixes GST-SA-2026-0001, CVE-2026-1940.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4854
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10903>

2026-01-12 13:48:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Define maximum chunk size in a single place
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10903>

2026-01-12 13:41:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Use GST_ROUND_UP_2() in two more places instead of a manual implementation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10903>

2026-01-12 13:22:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Use unsigned integers for data sizes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10903>

2026-01-12 13:21:48 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Remove pointless duplicated GST_ROUND_UP_2()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10903>

2026-02-08 16:01:21 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/rtp/gstrtpqdmdepay.c:
	  rtpqdm2depay: error out if anyone tries to use this element
	  Forbid use of this element and comment out all processing code.
	  There is no plausible reason this code should ever be executed in 2026
	  seeing that this was a streaming format produced by Darwin Streaming Server
	  ca 2009 which hasn't been in active use for well over a decade.
	  We simply error out for now as defensive measure and will remove the
	  element entirely in the next release cycle.
	  The processing functions are hairy and rather tedious to fix without
	  at least some sample stream at hand.
	  If anyone actually does have a legitimate need for this element and
	  can provide a sample streams, we will happily implement a depayloader
	  in Rust.
	  Fixes ZDI-CAN-28850, ZDI-CAN-28851, ZDI-CAN-28851, ZDI-CAN-28850, GST-SA-2026-0008.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4903
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4890
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10897>

2026-01-22 08:13:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix out-of-bounds read when parsing PlayReady DRM UUIDs
	  Fixes GST-SA-2026-0002.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4865
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10890>

2026-01-28 13:55:28 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson.build:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudioelement.h:
	* sys/osxaudio/gstosxaudiosink.c:
	* sys/osxaudio/gstosxaudiosink.h:
	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxaudiosrc.h:
	* sys/osxaudio/gstosxcoreaudio.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	* sys/osxaudio/gstosxcoreaudiocommon.c:
	* sys/osxaudio/gstosxcoreaudiocommon.h:
	  osxaudio: Stop using HAVE_IOS, use TARGET_OS_* macros instead
	  With osxaudio, all the ifdefs are macOS and !macOS, so that
	  automatically adds preliminary support for tvOS, watchOS, and visionOS
	  here.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10807>

2026-01-28 11:45:20 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/meson.build:
	* ext/qt/meson.build:
	* ext/qt6/meson.build:
	* gst/deinterlace/meson.build:
	* meson.build:
	* sys/osxaudio/meson.build:
	* sys/osxvideo/meson.build:
	  meson: Deprecate `system = 'ios'` in cross files, use subsystem
	  Since we require Meson 1.4.0, we can now use subsystem names (added in
	  1.2.0).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10807>

2023-04-02 16:53:45 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* scripts/dist-common-files.py:
	  modules: dist common files from monorepo root
	  Less noise when making releases, and just need to maintain one copy.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10822>

2026-02-15 17:45:19 +0000  Tim-Philipp Müller <tim@centricular.com>

	* README.md:
	  modules: remove subproject README.md from git
	  Will be added to the tarballs based on the monorepo README on dist.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10822>

2026-02-15 17:20:59 +0000  Tim-Philipp Müller <tim@centricular.com>

	* RELEASE:
	  modules: remove RELEASE from git, will be generated from template on dist
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10822>

2026-02-15 15:02:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	  modules: Remove NEWS from git which is generated from full release notes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10822>

2026-02-05 16:15:34 +0300  Andrey Sidorov <asidorov@elvees.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Memory leak in gst_rtspsrc_close() when GST_RTSP_EEOF error occurs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10768>

2026-01-29 09:36:41 +0100  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtsp/gstrtspsrc.c:
	  gstrtspsrc: Set new mki in the encoder upon crypto update
	  Upon crypto update, only the new master key is set on the encoder. Consider also the newly
	  generated mki
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10756>

2026-01-04 21:10:37 +0100  Christian Gräfe <cgraefe83@gmail.com>

	* docs/gst_plugins_cache.json:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	  rtpptdemux/rtpssrcdemux: adapt klass "Demux" to "Demuxer"
	  - use recommended "Demuxer" instead of just "Demux"
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10729>

2026-01-06 09:59:21 +0100  Christian Gräfe <cgraefe83@gmail.com>

	* docs/gst_plugins_cache.json:
	  gst: also adapt author names in the gst_plugins_cache.json files
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10729>

2026-01-04 19:25:58 +0100  Christian Gräfe <cgraefe83@gmail.com>

	* gst/debugutils/breakmydata.c:
	* gst/debugutils/testplugin.c:
	  gst-plugins: fix author name: correct incomplete or wrong emails
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10729>

2026-01-04 19:27:12 +0100  Christian Gräfe <cgraefe83@gmail.com>

	* gst/rtp/gstrtph263pay.c:
	  rtph263pay: fix author name: where not separated by ','
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10729>

2026-01-04 19:02:19 +0100  Christian Gräfe <cgraefe83@gmail.com>

	* gst/videofilter/gstgamma.c:
	* sys/v4l2/gstv4l2mpeg4enc.c:
	* sys/v4l2/gstv4l2vp8enc.c:
	* sys/v4l2/gstv4l2vp9enc.c:
	  gst: fix author name: add missing closing angle bracket
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10729>

2026-01-04 18:34:47 +0100  Christian Gräfe <cgraefe83@gmail.com>

	* gst/effectv/gstop.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstradioac.c:
	* gst/effectv/gstripple.c:
	* gst/effectv/gststreak.c:
	  gst-plugins-good: fix author name: Kentaro Fukuchi
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10729>

2026-01-29 13:21:06 +0100  Peter Stensson <petest@axis.com>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  rtpsource: Add locking for receive reports table
	  The hash table can be accessed simultaneously from multiple threads. The
	  rtpsessions rtcp thread will trigger collection of received report
	  stats when it's timeout triggers through the "on-sender-ssrc-active"
	  signal. Meanwhile rtpsession might process a new receive report and
	  modify the receive reports table in rtpsource. Since the session lock is
	  released before sending "on-sender-ssrc-active" signal there is no
	  protection for the receive report table in rtpsource.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10700>

2026-01-22 06:42:41 +0100  Tobias Koenig <tobias.koenig@kdab.com>

	* ext/qt6/qt6glwindow.cc:
	* ext/qt6/qt6glwindow.h:
	  qml6glsrc: Fix scraping of QQuickWindow content
	  When QQuickWindow emits the afterFrameEnd signal, the eglSwapBuffers
	  method has been invoked already for the current frame, so GL_BACK
	  does not contain the frame anymore that is displayed in the
	  QQuickWindow, but according to eglSwapBuffers some undefined data
	  (it happens to be the previous frame).
	  That causes the frame that qml6glsrc injects into the pipeline to
	  be always one frame later than what is visible in the QQuickWindow.
	  This patch connects to the QQuickWindow::afterRendering() signal
	  to make sure the glReadPixels() call is invoked before eglSwapBuffers
	  is invoked. To ensure that at this time the current frame is
	  completely rendered into GL_BACK, the QQuickWindow::begin/endExternalCommands()
	  is invoked to execute the pending commands from the scene graph renderer
	  in command buffer before we copy the GL_BACK content.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10678>

2026-01-20 15:31:17 +0100  Tobias Koenig <tobias.koenig@kdab.com>

	* ext/qt6/qt6glwindow.cc:
	  qml6glsrc: Fix rendering of scene with clipped items
	  If the scene of the QQuickWindow contains an Item with active
	  clipping ('clip: true'), the GL_SCISSOR_TEST is still active
	  when the frame buffer is copied over into gstreamer buffer
	  in Qt6GLWindow::afterFrameEnd().
	  Therefore only the content of that Item ends up in the
	  gstreamer buffer and not the whole window content.
	  This commit fixes it by resetting the OpenGL context to its
	  default state before the copy operation is done.
	  Since the afterFrameEnd callback is invoked after all the
	  QtQuick scene graph operations have been executed, there
	  is no need to restore the previous context state, the
	  scene graph renderer will setup the state from scratch
	  on the start of the next frame anyway.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10678>

2026-01-08 20:00:29 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	  wavpack: Update plugin docs cache
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 16:03:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackenc.h:
	  wavpackenc: Add support for S8/S16/S18/S20/S24/S32 and F32 samples
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 14:13:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackenc.c:
	  wavpackenc: Map buffer readwrite for channel reordering
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 14:09:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackenc.c:
	* ext/wavpack/gstwavpackenc.h:
	  wavpackenc: Extend caps a bit
	  Set width and depth like the parser does now, also set sample-type.
	  Allow for up to 4096 channels and maximum sample rate.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 14:01:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackdec.c:
	* ext/wavpack/gstwavpackdec.h:
	  wavpackdec: Output 18 and 20 bit as their corresponding formats
	  Instead of extending them to 24 bits.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 13:49:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: Allocate output buffer via the base class
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 13:47:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: Output 24 bit samples as actual 24 bit samples
	  Instead of extending them to 32 bits.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 13:35:18 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: Re-create wavpack decoder on caps changes
	  It doesn't support allow format changes in the middle of the stream and
	  continues decoding with the initial format, which leads to corrupted output.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4837
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 13:04:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: Allow up to 4096 channels and up to maximum sample rate
	  The previous values were lower than what the format allows.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 12:33:43 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/wavpack/gstwavpackdec.c:
	  wavpackdec: Set OPEN_NORMALIZE flag to normalize floating point samples into the [-1,1] range
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 12:25:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/gstwavpackparse.h:
	  wavpackparse: Correctly parse and advertise depth vs. width
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 11:59:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/gstwavpackparse.h:
	  wavpackparse: Include sample type (int / float / dsd) in the caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 11:49:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	* gst/audioparsers/gstwavpackparse.h:
	  wavpackparse: Sync flags with latest wavpack
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 11:47:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: Print hexadecimal numbers with leading zeroes for easier reading
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-05 11:46:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: Parse 32 bit sample rate and channel masks
	  And update template caps for the appropriate ranges, and also add missing +1 for
	  the number of channels.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10707>

2026-01-20 14:04:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: Push raw audio/video buffers downstream in reverse order if rate < 0
	  For compressed buffers the decoder base classes take care of this but for raw
	  data the demuxer has to do that as there is no other element downstream that
	  could potentially do it.
	  Downstream elements can't distinguish easily between the raw output of a decoder
	  and the non-reversed raw output of a demuxer due to a signalling design bug
	  (signalling for raw data is different than for compressed data) and this is the
	  best we can do for now.
	  Also make sure to set the discont flag correctly in this case.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10708>

2026-01-16 13:10:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Make sure to not output the same samples multiple times in reverse playback mode
	  By jumping from keyframe to keyframe backwards for one of the tracks and
	  outputting the overlapping samples of the other streams, it can happen that we
	  end up at the same samples as for the previous keyframe for the other tracks.
	  One example of this would be an intra-only video stream where the audio samples
	  have a considerably larger duration than a video frame. E.g. if audio samples
	  have 15 times the duration of a video frame, every audio sample would be output
	  15 times.
	  Apart from being inefficient, this also has the chance of breaking downstream
	  processing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10690>

2026-01-20 14:04:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Don't ignore flow return when pushing queued buffers downstream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10680>

2026-01-02 17:25:13 +0100  Robert Mader <robert.mader@collabora.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Support downstream pools with alignment requirements
	  Notably the VideoDmabufPool with the udmabuf allocator, allowing sinks
	  like the gtk4paintablesink or waylandsink to use dmabuf upload paths,
	  significantly improving performance in many cases.
	  Note that this only works when downstream pools can actually be used, which
	  is only the case for VP8, not VP9, for now.
	  Additional context:
	  VP9 currently unconditionally creates a internal pool in order meet the
	  requirements of the decoder (in the allocation callback). My attempts to
	  use the downstream pool instead failed because the decoder doesn't allow us
	  to set a custom stride (it's hardcoded in libvpx), preventing the use of
	  the udmabuf pool. There are still cases where a downstream pool could be
	  used and would be desirable - notably in glupload where using the internal
	  pool results in us using the raw uploader - but the complexity of the changes
	  didn't seem worth it for now.
	  VP8, OTOH, doesn't have a allocation callback but allocates buffers in libvpx,
	  and we always copy the frame to a buffer from a pool - so using a downstream
	  pool is trivial and already worked OOTB - just needed some changes to accept
	  downstream pools like the udmabuf one.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10666>

2026-01-21 11:37:03 -0800  Deepa Guthyappa Madivalara <deepa.madivalara@oss.qualcomm.com>

	* docs/gst_plugins_cache.json:
	  v4l2: update gst plugins cache
	  Updating gst plugins cache to resolve CI errors.
	  Sync gst_plugins_cache.json with CI-generated documentation cache.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10664>

2025-08-20 14:23:51 -0700  DEEPA GUTHYAPPA MADIVALARA <deepa.madivalara@oss.qualcomm.com>

	* sys/v4l2/ext/videodev2.h:
	* sys/v4l2/gstv4l2av1codec.c:
	* sys/v4l2/gstv4l2av1codec.h:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/meson.build:
	  v4l2: Add v4l2av1dec stateful decoder support
	  Introduce support for new pixelformat V4L2_PIX_FMT_AV1 which maps
	  to V4l2 AV1 stateful decoder. Implement necessary changes to
	  enable v4l2av1dec component.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10664>

2026-01-12 19:43:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Set the segment position to the start on EOS in reverse playback mode
	  This way it will be taken into account correctly when doing a non-flushing seek
	  afterwards. Otherwise it will have only the timestamp of the previous keyframe
	  that was inside the segment.
	  Also include a validate test for this.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10629>

2026-01-12 19:42:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Rename last mention of last_stop to position
	  0.10 is long gone.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10629>

2026-01-12 19:41:20 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
