=== release 1.29.1 ===

2026-03-22 14:56:37 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.29.1

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/11094>

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/11094>

2026-03-22 14:20:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* RELEASE:
	  modules: remove RELEASE from git, will be generated from template on dist

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/11094>

2020-09-13 03:04:53 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Timestamp outgoing buffers from mux, not incoming buffer
	  This ensures that the outgoing buffer has the right timestamp if we
	  pulled a backwards buffer forward.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1034>

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/11074>

2026-03-11 10:00:38 +0900  Shigeharu Kamiya <s_kamiya@toa.co.jp>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: add backchannel-http-method property for HTTP tunnel mode
	  Some ONVIF cameras reject backchannel RTP data sent base64-encoded via
	  the HTTP POST connection, expecting it as raw binary on the GET
	  connection instead. Add a backchannel-http-method property to select
	  which connection carries backchannel data (post or get), with automatic
	  fallback to the other method when the server closes the connection.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10999>

2026-03-16 16:37:34 +0200  Vivia Nikolaidou <vivia@ahiru.eu>

	* docs/gst_plugins_cache.json:
	* gst/flv/gstflvdemux.c:
	* gst/flv/gstflvdemux.h:
	  flvdemux: Make no-more-pads threshold configurable
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11042>

2026-02-23 11:41:29 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* tests/check/meson.build:
	* tests/check/pipelines/isomp4-vpcc.c:
	* tests/check/pipelines/vp9-codecprivate.c:
	  tests: add vp9 CodecPrivate and vpcC pipeline coverage
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10875>

2026-02-23 11:35:39 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Now parsing vp9 CodecPrivate when present
	  - Using new VPX helpers from codec-utils to populate caps.
	  - Allows initial vp9 decoder configuration without depending on parsing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10875>

2026-02-23 11:38:01 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* gst/matroska/matroska-ids.c:
	* gst/matroska/matroska-ids.h:
	* gst/matroska/matroska-mux.c:
	  matroskamux: adding vp9 CodecPrivate data to file when possible
	  - matroska-ids.[ch]: New ids and utilities to help matroskamux and
	  matroskademux deal with vp9 CodecPrivate parse/validation.
	  - matroska-mux.c: Now adding vp9 CodecPrivate when caps have appropriate
	  fields.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10875>

2026-02-23 11:34:51 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* gst/isomp4/qtdemux.c:
	  isomp4: switch qtdemux to VPX caps format helper
	  The original logic to convert vpcC values to caps fields was moved
	  to codec-utils so it can be shared with matroskademux when it
	  parses vp9 CodecPrivate.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10875>

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/10590>

2026-02-05 14:16:08 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* gst/isomp4/gstqtmux-doc.c:
	  qtmux: Add AV1 compatibility note in documentation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10704>

2026-01-30 16:59:14 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* tests/check/meson.build:
	* tests/check/pipelines/av1c-qtmux-qtdemux.c:
	* tests/files/README.av1c.md:
	* tests/files/av1c_min.mp4:
	* tests/files/av1c_two_entries.mp4:
	* tests/files/av1c_with_obu.mp4:
	  tests: qtmux, qtdemux: Add AV1 av1C round-trip tests
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10704>

2026-02-03 17:16:25 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmux.h:
	  qtmux: allow AV1 caps renegotiation and warn on unparsed input
	  Treat AV1 like H.264/H.265/H.266 during caps reconfiguration by allowing
	  format-level field changes (for example profile/level/tier-related updates)
	  without failing renegotiation.
	  Also warn once per pad when AV1 caps are not marked parsed=true. In that
	  case qtmux may have insufficient information to build a reliable av1C from
	  caps alone, so users should insert av1parse upstream.
	  This warning is intentionally advisory: qtmux cannot use GstAV1Parser directly
	  to derive missing configuration because that parser lives in -bad while qtmux
	  is in -good.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10704>

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/11049>

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/11014>

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/11044>

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/10997>

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/11008>

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/11004>

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.
	  This commit also adds a validate-flow test simply recording raw WebVTT
	  subtitles.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4954
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/11017>

2026-03-11 08:35:46 +0100  Ognyan Tonchev <ognyan@axis.com>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	* tests/examples/rtsp/test-client-managed-mikey.c:
	  test-client-managed-mikey: Preserve ROC upon key update
	  when using client-managed MIKEY KeyMgmt.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10980>

2026-02-06 22:01:06 +0700  sergey radionov <rsatom@gmail.com>

	* ext/qt6/meson.build:
	  qt6: fixed build for Android
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10781>

2026-02-14 21:29:47 +0100  Robert Mader <robert.mader@collabora.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Update and simplify decide_allocation
	  Following the av1dec example. Calling into the base class *after*
	  enabling the VIDEO_META and VIDEO_ALIGNMENT options makes the decoder
	  compatible with the new VideoDmabufPool.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10869>

2026-02-22 02:58:13 +0100  Robert Mader <robert.mader@collabora.com>

	* ext/vpx/gstvpxdec.c:
	  vpxdec: Simplify decide_allocation
	  The only relevant parts the implementation does over the base class
	  is enabling the VIDEO_META and VIDEO_ALIGNMENT options on the pool.
	  Simplify it accordingly, following the av1dec example.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10869>

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/10959>

2026-03-01 11:10:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtpopusdepay.c:
	* gst/rtp/gstrtpopuspay.c:
	  rtp: Lower Opus (de)payloader ranks to SECONDARY
	  To be in sync with all the others: none are > SECONDARY.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10940>

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/10951>

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/10920>

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/10900>

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/10900>

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/10900>

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/10900>

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/10900>

2026-02-08 16:09:04 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtp.c:
	* gst/rtp/gstrtpqdmdepay.c:
	* gst/rtp/gstrtpqdmdepay.h:
	* gst/rtp/meson.build:
	  rtpqdm2depay: remove element
	  There is no plausible reason anyone should need this element 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.
	  It is a maintenance burden and a security liability and there's no
	  good reason to keep it around.
	  Fixes ZDI-CAN-28850, ZDI-CAN-28851, CVE-2026-3083, CVE-2026-3085, GST-SA-2026-0008.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10886>

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/10886>

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/10880>

2026-01-29 10:21:03 -0700  Olivier Crête <olivier.crete@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_dump.c:
	* gst/isomp4/qtdemux_dump.h:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: Handle 'pict' tracks as if they were video tracks
	  Image tracks are sequences of timed images, so for all practical
	  purposes, they're video.
	  They are defined by ISO/IEC 23008-12 High Efficiency Image File Format (HEIF)
	  Also include a matching validate test.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10688>

2026-01-21 21:19:17 +1100  Jan Schmidt <jan@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-demux.h:
	* gst/matroska/matroska-parse.c:
	* gst/matroska/matroska-parse.h:
	* gst/matroska/matroska-read-common.c:
	* gst/matroska/matroska-read-common.h:
	  matroskademux: Build dynamic seek index if needed
	  when a stream doesn't have a cues table, build a seek
	  index dynamically and use it for seeking, to support
	  efficient reverse playback and faster seeks on
	  regions we've already parsed.
	  To support this, the demuxer will dynamically track
	  a set of seek_ranges that have been indexed.
	  If there is a cues table, it is used directly and
	  no dynamic index is tracked.
	  When traversing through regions that have not been
	  parsed, reverse playback will step back through clusters
	  and index them
	  This is useful for playing and scrubbing around
	  files that were written with streamable=true,
	  or that have not yet finished writing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10642>

2026-02-05 19:41:49 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/matroska-demux.c:
	* gst/matroska/matroska-ids.h:
	  matroska-demux: parse TrackLanguageBcp47 tags
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10766>

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/10667>

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/10667>

2026-02-10 08:54:48 +0100  Ognyan Tonchev <ognyan@axis.com>

	* tests/examples/rtsp/test-client-managed-mikey.c:
	  test-client-managed-mikey: Fix parsing of auth
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10783>

2026-01-23 10:14:12 +0100  Tobias Koenig <tobias.koenig@kdab.com>

	* docs/gst_plugins_cache.json:
	* ext/qt6/gstqml6glsrc.cc:
	* ext/qt6/gstqml6glsrc.h:
	* ext/qt6/gstqt6navigation.cc:
	* ext/qt6/gstqt6navigation.h:
	* ext/qt6/meson.build:
	  qml6glsrc: Add support to handle navigation events
	  Add support to qml6glsrc to handle mouse, keyboard and touch
	  navigation events, by converting them to corresponding QEvents
	  and injecting them into the QQuickWindow.
	  Since that might have security implications, the functionality
	  is guarded by a new 'view-only' property, which is 'true' by default.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10733>

2026-02-06 09:17:31 +0100  Ognyan Tonchev <otonchev@gmail.com>

	* tests/examples/rtsp/test-client-managed-mikey.c:
	  test-client-managed-mikey: Allow user to select auth method
	  Add support to select auth method. Example:
	  hmac-sha1-80 or null (the latter one shall be set for AEAD ciphers like
	  aes-128-gcm)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10745>

2026-01-29 09:41:15 +0100  Ognyan Tonchev <ognyan@axis.com>

	* tests/examples/rtsp/test-client-managed-mikey.c:
	  test-client-managed-mikey: Allow user to select cipher
	  Add support to select cipher for the SRTP encryption and authentication:
	  Example:
	  'aes-128-icm' or 'aes-256-icm' in combination with 'hmac-sha1'
	  or 'aes-128-gcm' or 'aes-256-gcm' which are authenticated encryption algorithms
	  Also improve option parsing
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10745>

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/10759>

2026-01-28 16:42:33 -0500  Daniel Morin <daniel.morin@collabora.com>

	* gst/isomp4/fourcc.h:
	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux_types.c:
	  qtdemux: adding support for bayer to mp4(uncv)
	  - ComponentPatternDefinitionBox added to describe bayer format
	  - Use 2x2 pattern, with no gain (=1)
	  - register 'uncv, uncC, cpat, cmpd' to quicktime node types
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10658>

2026-02-05 15:50:06 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/isomp4/meson.build:
	  qtmux: include gstqtmux-doc.c in doc_sources
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4887
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10763>

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/10744>

2025-12-09 15:42:07 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstdtmfsrc.h:
	* gst/dtmf/gstrtpdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.h:
	  dtmfsrc: expose properties for controlling minimum durations
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4775
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10334>

2025-12-09 15:40:15 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/dtmf/gstrtpdtmfdepay.c:
	  rtpdtmfdepay: remove unused define
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10334>

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/10728>

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/10728>

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/10728>

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/10728>

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/10728>

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/10728>

2026-01-20 14:41:11 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Extract vp9 level from vpcC into caps using codec-utils
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10538>

2026-01-20 14:40:31 -0500  Dominique Leroux <dominique.p.leroux@gmail.com>

	* gst/isomp4/atoms.c:
	* gst/isomp4/atoms.h:
	* gst/isomp4/gstqtmux.c:
	  qtmux: Use codec-utils to generate vpcC and av1c instead of local code
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10538>

2025-10-16 16:30:04 +0200  Edward Hervey <edward@centricular.com>

	* gst/deinterlace/gstdeinterlace.c:
	  deinterlace: Allow allocation queries to pass through in mixed mode
	  With mixed interlacing mode, we need to wait until we get buffers to know
	  whether they can be forwarded as-is or whether they need to be deinterlaced.
	  In such cases, we forward the allocation query downstream so that upstream has a
	  chance to get an optimal bufferpool (and other video information) to decode
	  directly into if it's progressive.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10579>

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/10682>

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/10571>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

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/10496>

2025-12-27 19:17:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/spectrum/gstspectrum.c:
	  spectrum: Return NOT_NEGOTIATED if transform_ip() is called without caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10461>

2025-12-27 19:16:35 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/level/gstlevel.c:
	* gst/level/gstlevel.h:
	  level: Port from GstBaseTransform to GstAudioFilter
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4830
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10461>

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/10683>

2025-12-19 17:31:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Include start/end timecodes in fragment-opened/closed messages
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10420>

2021-01-24 12:21:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Correctly parse RTSP 2.0 RTP-Info headers
	  These are actually different from RTSP 1.0 ones and failed to parse
	  before. As gst-rtsp-server produces RTSP 1.0 RTP-Info headers, still
	  handle parsing them for RTSP 2.0.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3064
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8571>

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/10548>

2026-01-26 09:00:11 +0000  ArokyaMatthewNathan <arokyamatthewnathan@gmail.com>

	* gst/alpha/gstalphacolor.c:
	  gstalphacolor: use G_GNUC_UNUSED for unused parameters
	  Mark unused function arguments with G_GNUC_UNUSED macro instead of
	  casting to void, to suppress -Wunused-parameter warnings.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10600>

2026-01-20 15:51:43 +0100  Tobias Koenig <tobias.koenig@kdab.com>

	* ext/qt6/gstqml6glsrc.cc:
	* ext/qt6/qt6glwindow.cc:
	* ext/qt6/qt6glwindow.h:
	  qml6glsrc: Simplify implementation of Qt6GLWindow
	  The Qt6GLWindow class doesn't have to inherit from QQuickWindow,
	  since it doesn't provide any window-specific functionality that
	  is used by qml6glsrc.
	  Let Qt6GLWindow inherit from QObject instead and adapt the order
	  of constructor parameters to the Qt coding style (parent parameter
	  is always last).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10573>

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/10570>

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/10572>

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/10572>

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/10442>

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/9892>

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/9892>

2026-01-26 13:15:40 +0100  Matus Gajdos <matuszpd@gmail.com>

	* gst/apetag/gstapedemux.c:
	  apedemux: Add support for cover art tags
	  Try and extract `Cover Art (Front)` and `Cover Art (Back)` APE tags into
	  GST_TAG_IMAGE.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10603>

2026-01-27 15:51:14 +0100  Matus Gajdos <matuszpd@gmail.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Add support for ID3 tags
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10610>

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/10534>

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/10534>

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

	* gst/isomp4/qtdemux.c:
	  qtdemux: Improve debug output around seek event handling
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10534>

2026-01-27 18:25:19 +0100  Piotr Brzeziński <piotr@centricular.com>

	* ext/adaptivedemux2/meson.build:
	  meson: Fix libxml2 not building due to wrong option type
	  'python' was moved from a boolean to a feature a few months ago and
	  4f4260dbe3489699aba0a724a3d55020666a0d6a pulled that in on our side.
	  Notably, this was causing adaptivedemux2 to not build on my system.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10615>

2026-01-27 18:36:19 +0000  Tim-Philipp Müller <tim@centricular.com>

	* README.md:
	* RELEASE:
	* meson.build:
	  Back to development in main branch after 1.28.0
	  - Track orc main branch
	  - Track gst-plugins-rs main branch
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10616>

=== release 1.28.0 ===

