=== release 1.26.11 ===

2026-03-10 13:08:25 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.11

2026-03-05 15:35:15 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	  opusenc: Use correct memcpy() size when copying Vorbis channel positions
	  The size of the destination (64 channels) was used but the source is
	  smaller (8 channels). While copying garbage into the destination is not so much
	  of a problem as only the actually used number of channels is going to be read
	  later, this can potentially lead to a segfault.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4948
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10985>

2026-02-03 07:30:26 +0900  jeongmin kwak <jeongmin.kwak@lge.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	* tests/check/libs/rtp.c:
	  rtcp: Fix buffer overread in SDES packet parsing
	  Add bounds checking in gst_rtcp_packet_sdes_next_entry() and
	  gst_rtcp_packet_sdes_get_entry() to prevent reading beyond packet
	  boundaries when parsing malformed SDES items.
	  Both functions now return FALSE when attempting to read past the
	  packet length, preventing potential buffer overread vulnerabilities.
	  tests: Add regression test for SDES buffer overread vulnerability
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10972>

2026-03-03 16:25:39 +0900  jeongmin kwak <jeongmin.kwak@lge.com>

	* gst-libs/gst/rtp/gstrtpbuffer.c:
	* tests/check/libs/rtp.c:
	  rtpbuffer: Add validation for CSRC list length
	  Validate that the CSRC list does not extend beyond the packet boundary
	  as required by RFC 3550. A malformed RTP packet with a CSRC count that
	  would cause the CSRC list to extend beyond the available packet data
	  could lead to out-of-bounds memory access.
	  The validation follows the same pattern used for padding validation
	  (line 430), ensuring consistency with existing security checks.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10958>

2026-01-28 11:50:49 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtsp/gstrtsprange.c:
	* gst-libs/gst/rtsp/gstrtsptransport.c:
	* gst/subparse/gstsubparse.c:
	* gst/subparse/qttextparse.c:
	* gst/subparse/samiparse.c:
	  base: Fix a couple of new const-ness warnings around strstr() usage
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4871
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10657>

2026-01-09 16:42:52 +0100  François Laignel <francois@centricular.com>

	* tests/check/elements/inputselector.c:
	  input-selector: add test eos for remaining non active pad
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10622>

2026-01-06 16:43:55 +0100  François Laignel <francois@centricular.com>

	* tests/check/elements/inputselector.c:
	  input-selector: add pad release stress test
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10622>

2025-12-28 21:19:54 -0500  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/inputselector.c:
	  input-selector: test: Fix memory leaks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10622>

2025-12-28 18:12:54 -0500  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/inputselector.c:
	  input-selector: test: Don't use g_assert() in tests
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10622>

2025-12-28 17:51:38 -0500  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/inputselector.c:
	  input-selector: test: increase delay when running under valgrind
	  When running under valgrind we could get into a loop where we'd
	  read only from one source since we'd switch back too quickly.
	  Test would eventually timeout with the count from one source being
	  above 30000 and the other never reaching the minimum.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10622>

2025-12-28 17:45:32 -0500  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/inputselector.c:
	  input-selector: test: quieten output
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10622>

2025-12-01 19:32:54 +0100  François Laignel <francois@centricular.com>

	* tests/check/elements/inputselector.c:
	* tests/check/meson.build:
	  input-selector: add stress test
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10622>

2026-02-11 19:44:34 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: Correctly check that enough RGB palette data is available
	  This can otherwise overflow and result in out-of-bounds reads/writes.
	  Fixes GST-SA-2026-0004, ZDI-CAN-28854, CVE-2026-2921.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4901
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10929>

2026-02-03 14:34:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Don't directly forward gap events if the decoder is not drained
	  Instead queue it up for forwarding before the next video frame. While this is
	  not ideal, it makes sure that the gap event does not go ahead of previous
	  buffers if the decoder has any queued internally, which could break timestamp
	  tracking downstream.
	  This fixes a regression introduced in 62cb08404b9bd055c45cc1d5b71141e927ad748a.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10751>

2026-02-03 14:13:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Move gap event handling into a helper function
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10751>

2026-02-03 13:55:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	  audiodecoder: Only forward gap events immediately if no frames are pending
	  Otherwise the gap event would go ahead of the pending frames, breaking
	  buffer/event serialization.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10751>

2026-01-06 11:41:27 +0900  jeongmin kwak <jeongmin.kwak@lge.com>

	* gst-libs/gst/rtsp/gstrtsptransport.c:
	  rtsp: Validate transport parameter parsing in RFC 2326
	  The layers, ttl, and ssrc parameters were parsed with strtoul()
	  but did not validate whether parsing succeeded, causing invalid or
	  empty values to be silently accepted as zero in violation of RFC 2326.
	  Replace with g_ascii_string_to_unsigned() for proper validation with
	  automatic range checking (ttl: 0-255) and overflow detection.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10718>

2026-01-30 13:07:48 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst/playback/gstplaysink.c:
	  playsink: unref color balance channels with g_object_unref()
	  Since they are not GstObject but plain GObjects, the unref function has to be
	  g_object_unref(). Otherwise it might generate problems with tracers.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10711>

2025-12-12 06:30:48 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Handle recovery from temporary reordered output
	  The code previously assumed that if there was a single reordered output
	  frame, then it was wrong forever.
	  Sadly, it could just be a temporary issue (maybe even just a single frame) and
	  wouldn't have a chance from recovering.
	  Instead, check if we received enough "ordered" frames from the subclass and
	  switch back to normal behaviour
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10693>

2025-04-03 14:52:49 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Fix switch to smaller collections
	  If an input stream goes away without EOS, we will end up with a case where the
	  number of slots won't match the number of stream in the collection (until that
	  stream and slot really gets removed).
	  But we still want to detect when all streams of that (smaller) collection are
	  present so we can switch to it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10675>

2025-03-26 16:07:18 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Improve handling collection change on existing pad
	  A check was already present for checking if an incoming stream belonged to the
	  current collection. This was added in
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10109
	  But it only focused on the collection on the current pad, and not on the
	  aggregated collection.
	  This modification actually checks against the aggregated (and potentially
	  different) input collection, and also updates it accordingly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10640>

2025-03-26 16:01:02 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Ensure input->collection is always coherent
	  If we want to "unset" the current input collection (by providing NULL) it
	  wouldn't actually unset it and we would end up with a dangling pointer instead
	  of NULL (i.e. no collection yet for that input)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10640>

2026-01-14 23:12:32 +0100  Ruben Gonzalez <rgonzalez@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefindfunctions: Promote y4m_typefind
	  Related to https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4848
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10631>

2026-01-06 10:40:09 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstplaybin3.c:
	* gst/playback/gsturidecodebin3.c:
	  playbin3: Move locking down to uridecodebin3
	  uridecodebin is always present in playbin3, the locking should be done within
	  uridecodebin instead
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4822
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10517>

2025-12-30 18:11:43 -0500  Doug Nazar <nazard@nazar.ca>

	* ext/ogg/gstoggdemux.c:
	* ext/ogg/gstoggelements.h:
	* ext/ogg/gstoggmux.c:
	  oggmux: Fix crash when debug logging enabled
	  The muxer uses the ogg_stream functions however that file uses the
	  demuxer debug categories. Ensure they are also initialized.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10482>

2025-12-30 17:57:37 -0500  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/textoverlay.c:
	  textoverlay: tests: Fix race clearing element and debug printing
	  It was possible to unblock the main thread and unref the pipeline
	  before we had exited the gst_element_set_state() in the thread.
	  With GST_DEBUG enabled it could crash trying to print the element name.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10482>

2025-12-31 11:58:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin3.c:
	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Fix docs for the select-stream signal
	  It's GstURIDecodeBin3 and not GstURIDecodebin3.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10481>

2025-12-30 18:53:26 -0500  Doug Nazar <nazard@nazar.ca>

	* ext/pango/gstbasetextoverlay.c:
	* gst/audiomixer/gstaudiointerleave.c:
	* gst/overlaycomposition/gstoverlaycomposition.c:
	* gst/playback/gstsubtitleoverlay.c:
	  gst: Properly unref pad template caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10479>

2025-12-30 17:53:22 -0500  Doug Nazar <nazard@nazar.ca>

	* tests/check/libs/videoencoder.c:
	  tests: Fix several memory leaks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10478>

2025-12-25 15:49:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.10

=== release 1.26.10 ===

2025-12-25 15:44:26 +0100  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.10

2025-12-18 19:58:29 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstparsebin.c:
	  Revert "parsebin: Don't expose unfixed caps pads without stream info"
	  This reverts commit 6628f5644e2c82da7f93012f9f4a8e08d9a4b3e1.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10405>

2025-12-18 19:56:42 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstparsebin.c:
	  Revert "parsebin: Clear old stream when retargeting pad"
	  This reverts commit d26eb95aec95e329c443ac238a13dc024f39dd0f.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10405>

2025-12-11 21:54:53 -0500  Aaron Boxer <aaron.boxer@collabora.com>

	* gst/playback/gstplaybin3.c:
	  playbin3: send GST_EVENT_SELECT_STREAMS event to collection source
	  this ensures that the source receives the event, even if pipeline is not
	  linked
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10394>

2025-12-16 23:47:48 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Improve debug logging
	  Only log that we're storing caps on a GstStream if actually
	  doing it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10397>

2025-12-16 23:11:23 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Don't expose unfixed caps pads without stream info
	  Pads with unfixed caps can be exposed early, but not before
	  they've at least provided a stream-start event for building
	  a fallback collection. Accordingly, defer the exposure and
	  check again when seeing a stream-start event.
	  Fixes #4756
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10397>

2025-12-16 23:09:02 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Clear old stream when retargeting pad
	  When extending the parsebin chain and retargeting the
	  current end pad, clear the previous stream info, or
	  the wrong stream info might get added to a fallback
	  collection
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10397>

2025-12-13 17:00:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusenc.c:
	* ext/opus/meson.build:
	  opusenc: Use the surround multistream encoder for the Vorbis channel mapping family
	  It provides additional surround features. For all other channel mapping families,
	  continue using the normal multistream encoder as it allows to provide the
	  channel configuration values.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10376>

2025-12-13 13:50:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusdec.c:
	* ext/opus/gstopusdec.h:
	  opusdec: Only call into the channel reordering functions if actually needed
	  And simplify code a bit.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10376>

2025-12-13 13:42:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopuscommon.c:
	* ext/opus/gstopuscommon.h:
	* ext/opus/gstopusenc.c:
	* ext/opus/gstopusenc.h:
	  opusenc: Simplify Vorbis channel layout mapping code and fix 7.1 layout
	  Don't set up complicated mapping tables but instead use the default mapping
	  tables as used by libopus. While this requires reordering from the GStreamer
	  order to the Vorbis order in some situations, it's a better choice because it
	  allows using the surround multistream encoder which has additional surround
	  support that saves bandwidth.
	  Also various specs and implementations only assume the "default" positions,
	  and e.g. for the MPEG-TS mapping this has short forms in the signalling. And
	  again, various implementations (e.g. ffmpeg) only support the short forms.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4794
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10376>

2025-05-21 13:04:07 -0400  F. Duncanh <fduncanh@gmail.com>

	* gst-libs/gst/gl/cocoa/gstglwindow_cocoa.m:
	  glwindow_cocoa: fix window not closing (w/o user window handle)
	  A user-supplied window handle (external_view) becomes the superView
	  of internal_view, which is closed with [view removeFromSuperview].
	  This fails silently if external_view = NULL (no handle supplied).
	  Call [win_internal_id close] in this case. Fixes #4432.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10361>

2025-12-10 15:11:23 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst-libs/gst/tag/gsttagdemux.c:
	  tagdemux: propagate seek event seqnum to upstream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10359>

2025-12-10 14:54:14 +0100  Piotr Brzeziński <piotr@centricular.com>

	* ext/pango/gstbasetextoverlay.c:
	  basetextoverlay: Don't negotiate if caps haven't changed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10357>

2025-12-08 13:20:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Don't assume the ALLOCATION query contains a pool
	  The subclass' `decide_allocation()` implementation might not provide any.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10347>

2025-12-05 17:22:45 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/gstglutils.c:
	  gstglutils: Round height according to the subsampling
	  When calculating the plane size, its important to round up the height according
	  to its subsampling. Otherwise the offset is too short and the UV get shifted
	  vertically.
	  Fixes #4783
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10345>

2025-12-05 17:21:02 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/gstglutils.c:
	  gstglutils: Refactor plane data size calculation
	  None of the work we do at the top is needed for the tiled formats. Handle the
	  tiled format first and return, then handle the other formats.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10345>

2025-12-08 09:32:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: Add support for FLAC in mime codec strings
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4789
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10327>

2025-12-08 09:31:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: Add support for AC4 in mime codec strings
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10327>

2025-12-08 09:31:06 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: Update mime codec string getter with codecs in the parser
	  Specifically, add AC3, EAC3, TTML and WebVTT.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10327>

2025-12-05 11:16:29 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideoaggregator.c:
	  videoaggregator: Fix decide allocation fallback
	  Fallback to generic pool, even if the change to the config was validated. Pools
	  may have other restrictions that may render them unusable.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10304>

2025-12-03 19:00:49 +0100  Robert Mader <robert.mader@collabora.com>

	* gst/videotestsrc/gstvideotestsrc.c:
	  testsrc: Enable POOL_OPTION_VIDEO_ALIGNMENT if possible
	  If supported by the pool and only if downstream supports VIDEO_META.
	  This is in line with e.g. gst_ffmpegviddec_decide_allocation() and
	  helpful with e.g. the udmabuf allocator.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10304>

2025-12-03 17:51:01 +0100  Robert Mader <robert.mader@collabora.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Improve pool_set_config() handling
	  set_config() can fail for other reasons than
	  `pool_config_validate_params()` failing, in which case we currently
	  would not try to fall back to the fallback pool.
	  Also ensure to get the config of the new pool instead of trying to set
	  the previously failed config again.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10304>

2025-12-05 14:11:01 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/allocators/gstfdmemory.c:
	  fdmemory: Fix size calculation when sharing
	  When sharing a memory with an offset, the offset is relative to the immediate
	  parent offset. Meaning we should be reducing the size accordingly.
	  Reported-by: Rouven Czerwinski <rouven.czerwinski@linaro.org>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10309>

2025-12-01 17:32:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.9

=== release 1.26.9 ===

2025-12-01 17:27:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.9

2025-12-01 10:49:02 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Protect NULL dereference
	  If we *do* end up in this code path, notify it
	  See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4626
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10253>

2025-11-20 12:49:12 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/allocators/gstdrmdumb.c:
	  allocators: drmdumb: Keep dmabuf mapped
	  The dumb buffer are used by waylandsink to allow copying the incoming frames
	  into a memory allocation directly supported by the GPU or display controller.
	  These buffers are pooled and avoiding to recreate the memory address mapping
	  everything improves the performance.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10164>

2025-11-20 12:46:49 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gl/gstgldownloadelement.c:
	  gldownload: Keep dmabuf mapped
	  The DMABuf support in gldownload allows forcing the GPU to write into a format
	  GStreamer understand. It is possible that this mechanism would be used for CPU
	  access were avoiding to recreate the address mapping is preferred.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10164>

2025-12-01 12:39:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst/playback/gstrawcaps.h:
	  playback: Consider meta/x-klv, meta/x-id3 and meta/x-st-2038 as raw formats
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10251>

2025-12-01 12:42:23 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: descriptions: Add ST-2038 to the codec descriptions
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10251>

2025-11-21 16:07:18 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* tests/check/libs/pbutils.c:
	  pbutils: Add a test case for another profile, tier, level
	  Add a test case to verify MIME codec string for H.265
	  profile, level, and tier data
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10183>

2025-11-17 15:40:05 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  pbutils: Fix bit shifting when generate hevc mime codec string
	  According to the HEVCDecoderConfigurationRecord configuration,
	  profile_space and tier_flag should be corrected.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10183>

2025-11-19 14:18:32 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepayload: Add missing unlocks
	  Add missing unlocks when retrieving information from the RTP header.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10157>

2025-09-23 22:32:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/alsa/gstalsadeviceprovider.c:
	  alsadeviceprovider: Fix device name leak
	  Also use g_free for string allocated with g_strdup*. This usually
	  doesn't matter, but can matter when mixing code compiled with
	  different libc.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10228>

2025-11-26 16:07:26 +0100  Stéphane Cerveau <scerveau@igalia.com>

	* gst-libs/gst/video/gstvideoencoder.c:
	* gst-libs/gst/video/gstvideoencoder.h:
	  videoencoder: fix warning of uninitialized buffer
	  Move the error check before the buffer reference to avoid using
	  an uninitialized buffer variable when ret is not GST_FLOW_OK.
	  Also fix whitespace in header documentation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10215>

2025-11-21 09:19:58 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  audiovisualizer: Remove unnecessary unmap call for adapter
	  The unmap function for the adapter is unnecessary because
	  gst_adapter_map() is not called anywhere in the chain function.
	  This patch removes the redundant unmap call to simplify
	  the code and avoid confusion.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10184>

2025-11-20 15:16:00 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  audiovisualizer: Change buffer access to go through adapter
	  Since ownership of the input buffer is transferred to the adapter
	  via gst_adapter_push, modify the buffer copy process to access
	  the buffer through the adapter.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10184>

2025-11-20 14:46:33 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  audiovisualizer: Fix resource leak in chain function
	  The input buffer should be unreferenced when
	  a negotiation failure is detected in the chain function.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10184>

2025-11-19 13:42:01 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  audiovisualizer: Use break instead of goto for escape logic
	  Use break instead of goto to escape and prevent cases
	  where config_lock might remain unlocked.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10184>

2025-11-21 11:55:47 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play.c:
	  gst-play-1.0: fix printing of missing plugin details
	  It wouldn't print what plugin was missing, because element
	  messages were handled already in a different place.
	  Also use g_list_prepend() here since order doesn't matter.
	  See #4758
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10176>

2025-11-19 14:26:13 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Add missing unlock
	  The lock of urisourcebin should be released
	  when the creation of the parsebin element fails.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10158>

2025-08-12 16:42:25 +1000  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Clear previous collection on input
	  When receiving a stream-start with a stream that isn't in the
	  previous input stream-collection, clear the previous stream collection
	  and treat it the same as the very first stream start event. That makes
	  decodebin3 delay processing the stream-start and any subsequent events
	  until the stream-collection arrives.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10156>

2025-11-19 14:11:31 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* ext/gl/gstglstereosplit.c:
	  glstereosplit: Add missing unlock for exceptional case
	  The context_lock should be released when gst_gl_ensure_element_data fails.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10148>

2025-11-19 14:06:56 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* tools/gst-play.c:
	  gst-play: Add missing unlock for invalid track type
	  Add missing unlock in play_cycle_track_selection
	  of gst-play. The unlock should be added when
	  the function ends in an exceptional case.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10147>

2025-11-19 13:51:27 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst-libs/gst/gl/gstglbasesrc.c:
	  glbasesrc: Add unlock handling for non-negotiated cases
	  Add missing unlock when returning
	  not negotiated status in glbasesrc.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10140>

2025-11-18 15:17:05 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Add null check of play items in purge
	  Add null check for play items in purge_play_items
	  to ensure that play_item is not null.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10134>

2025-11-19 13:36:46 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* ext/gl/gstglfiltershader.c:
	  glfiltershader: Add missing unlock
	  Add missing unlock in glfiltershader
	  for an exceptional return case.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10133>

2025-11-17 17:42:15 +0900  Jaehoon Lee <jaehoon85.lee@lge.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Fix initial values of min_byte_level and min_time_level variables
	  The min_byte_level and min_time_level should be initialized to
	  apply the minimum values of byte_limit and time_limit.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10132>

2025-10-13 16:25:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
	  rtpbaseaudiopay: Consider RESYNC flag as discontinuity too
	  Previously the marker bit would only be set by the payloader if the DISCONT
	  buffer flag is set, however the depayloader would only set the RESYNC flag
	  whenever it receives a packet with the marker bit set.
	  To avoid unnecessary DISCONTs after the depayloader, also consider the RESYNC
	  flag inside the payloader but keep the depayloader behaviour as it was.
	  See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/736
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10066>

2025-11-05 15:20:51 +0900  JihoonLee <ejihoon.lee@lge.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: Fix memory leak in _create_shader
	  Added missing g_string_free() call to properly free the GString buffer
	  in error paths where the function returns early.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10065>

2025-11-10 17:30:54 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.8

=== release 1.26.8 ===

2025-11-10 17:22:05 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.8

2025-10-31 15:16:01 +0900  Jeehyun Lee <jeehyun.lee@lge.com>

	* gst-libs/gst/gl/egl/gstglcontext_egl.c:
	  gl/egl: fix memory leak in _print_all_dma_formats
	  Add missing g_free() calls for fmt_str returned by
	  gst_video_format_to_string() in _print_all_dma_formats().
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10013>

2025-11-06 09:30:13 +0100  Mazdak Farzone <mazdak@axis.com>

	* ext/opus/gstopusdec.c:
	  opusdec: Unref intersected caps when empty to avoid leaks
	  Add gst_caps_unref(tmp) in the else branch of both the 'channels' and 'rate'
	  preference blocks to correctly release the empty intersection.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10031>

2025-10-30 10:38:09 -0300  Carlos Bentzen <cadubentzen@igalia.com>

	* docs/plugins/gst_plugins_cache.json:
	* ext/gl/gstglcolorbalance.c:
	* gst-libs/gst/gl/gstglcolorconvert.h:
	* gst-libs/gst/gl/gstglupload.c:
	* tests/check/libs/gstglcolorconvert.c:
	* tests/check/libs/gstglupload.c:
	  gl: Support DMABuf passthrough with meta:GstVideoOverlayComposition
	  The GL elements in glsinkbin were rejecting passing through DMABufs when
	  the meta:GstVideoOverlayComposition feature is present, such as when
	  subtitleoverlay is in place. This is particularly visible when playing
	  back HDR10 content with subtitles, where Showtime was showing washed
	  out colors due to the lack of tone-mapping in the GL elements when
	  converting from BT2100-PQ to RGBA due to passthrough being disabled.
	  This extends MR !5948 by adding support for DMABuf caps with the
	  meta:GstVideoOverlayComposition feature when such caps can be negotiated
	  downstream, such as with gtk4paintablesink.
	  Example of pipeline that was previously converting to GLMemory and RGBA,
	  which now passes through DMABUFs:
	  filesrc location='p010_bt2100-pq_subtitles.mkv' ! matroskademux name=d \
	  d. ! queue ! h265parse ! vah265dec ! subtitleoverlay name=s \
	  d. ! queue ! s. \
	  s. ! glsinkbin sink=gtk4paintablesink
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10011>

2025-11-05 13:56:18 +0900  JihoonLee <ejihoon.lee@lge.com>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: Fix memory leak in gst_encoding_target_save
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10021>

2025-11-05 10:04:36 +0900  Jeehyun Lee <jeehyun.lee@lge.com>

	* gst-libs/gst/tag/id3v2frames.c:
	  id3: fix csets memory leak in string_utf8_dup
	  Free csets array when charset conversion loop completes without
	  finding a valid conversion to prevent memory leak.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/10015>

2025-10-31 15:54:36 +0900  Jeehyun Lee <jeehyun.lee@lge.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: fix memory leaks in gst_rtsp_connection_connect_with_response_usec
	  Fix request_uri memory leaks in connection retry loop and properly
	  handle connection_uri ownership transfer during redirects.
	  - Initialize request_uri to NULL
	  - Free request_uri at start of each loop iteration
	  - Transfer ownership correctly during redirect handling
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9983>

2025-10-31 10:01:40 +0900  amy ko <amy.ko@lge.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Free missing plugin details and return failure when plugin is not found
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9970>

2025-10-31 14:33:11 +0900  Jeehyun Lee <jeehyun.lee@lge.com>

	* ext/gl/gltestsrc.c:
	  gltestsrc: Fix memory leaks on shader creation failure
	  Free coord and plane_indices when color_shader or snow_shader
	  compilation fails to prevent memory leaks in error paths.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9966>

2025-10-27 14:15:11 +0000  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  pbutils: Don't throw critical for unknown mime codec
	  Avoid a critcal in gst_codec_utils_caps_from_mime_codec()
	  trying to append NULL caps when processing an unrecognised
	  codec string
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9926>

2025-10-14 18:34:39 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.7

=== release 1.26.7 ===

2025-10-14 18:25:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.7

2025-10-09 16:56:09 +0530  Taruntej Kanakamalla <tarun@centricular.com>

	* gst/videorate/gstvideorate.c:
	* tests/check/elements/videorate.c:
	  videorate: fix assert fail due to invalid buffer duration
	  In `drop-only` mode, when pushing an input buffer, the buffer duration
	  was always assumed valid i.e. `invalid_duration`  as FALSE. This was causing
	  an assert failure when first few buffers don't have the duration calculated yet
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2886
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9829>

2025-10-09 13:07:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	  discoverer: Mark gst_discoverer_stream_info_list_free() as `transfer full`
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9822>

2025-09-29 16:11:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tests/examples/gl/qt/mousevideooverlay/pipeline.cpp:
	* tests/examples/gl/qt/qglwidgetvideooverlay/pipeline.cpp:
	* tests/examples/gl/qt/qglwtextureshare/meson.build:
	* tests/examples/gl/qt/qglwtextureshare/qglrenderer.cpp:
	  qt: Fix building examples on macOS
	  GL/gl.h is an incorrect include on macOS, and the correct include is
	  already done by gstgl.
	  qt_current_nsopengl_context() wasn't being exported, leading to build
	  failures
	  Also add comments to gstqtglutility.cc to clarify dense #ifdef usage.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9768>

2025-09-23 22:21:46 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: Add channel reorder maps for 3 and 7 channel audio
	  The function gst_riff_wave_add_default_channel_mask() was failing for
	  3 and 7 channel as it was not supported. We now use
	  gst_audio_channel_get_fallback_mask() to get the
	  default channel masks and add proper reorder maps for these cases.
	  The fallback layout follows ALSA conventions exactly as before, we're
	  just now leveraging the existing GStreamer infrastructure instead of
	  duplicating the logic.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9744>

2025-09-20 20:23:06 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst-libs/gst/sdp/gstmikey.c:
	  sdp: proper usage of gst_buffer_append
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9720>

2025-09-17 20:52:41 -0700  Xavier Claessens <xclaessens@netflix.com>

	* gst-libs/gst/glib-compat-private.h:
	* tools/gst-device-monitor.c:
	  build: Fix build error with glib < 2.68
	  glib-compat-private.h is copied verbatim from -good into -base because
	  g_string_replace() is used in gst-device-monitor since 1.26.6:
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9634
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9708>

2025-09-14 13:19:28 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.6

=== release 1.26.6 ===

2025-09-14 13:13:58 +0100  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.6

2025-09-05 18:32:43 +0300  Marko Kohtala <marko.kohtala@gmail.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Add get_url and get_ip return value annotation
	  The returned GstRTSPUrl is the one owned by the GstRTSPConnection,
	  but missing annotation causes caller to free it. This leads to access
	  to freed memory and eventually a crash.
	  The returned IP address annotation is missing it can be NULL when
	  the connection is closed.
	  Fixes #3726
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9656>

2025-09-05 08:30:28 -0400  Xavier Claessens <xclaessens@netflix.com>

	* gst-libs/gst/video/gstvideometa.c:
	  videometa: Fix valgrind warning when deserializing video meta
	  Conditional jump or move depends on uninitialised value(s)
	  at 0xAC8E1C8: gst_video_meta_is_alignment_valid (gstvideometa.c:589)
	  by 0xAC8E1C8: gst_video_meta_validate_alignment (gstvideometa.c:603)
	  by 0xAC8F741: gst_video_meta_set_alignment (gstvideometa.c:655)
	  by 0xAC8F9A7: video_meta_deserialize (gstvideometa.c:308)
	  by 0xA56E503: gst_meta_deserialize (gstmeta.c:856)
	  by 0x1472CE60: gst_unix_fd_src_create (gstunixfdsrc.c:390)
	  by 0xABCEAA3: gst_base_src_get_range (gstbasesrc.c:2632)
	  by 0xABD1C39: gst_base_src_loop (gstbasesrc.c:2964)
	  by 0xA5B1DC6: gst_task_func (gsttask.c:399)
	  by 0x8423383: ??? (in /usr/lib/x86_64-linux-gnu/libglib-2.0.so.0.6400.6)
	  by 0x8422AE0: ??? (in /usr/lib/x86_64-linux-gnu/libglib-2.0.so.0.6400.6)
	  by 0x485C608: start_thread (pthread_create.c:477)
	  by 0x4AFC352: clone (clone.S:95)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9654>

2025-09-01 21:02:22 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/gl/meson.build:
	* ext/opus/meson.build:
	* ext/pango/meson.build:
	* ext/vorbis/meson.build:
	  meson: Convert all remaining fallback: usages to [provide]
	  Only commonly-used plugin deps like pango, orc, openh264, libvpx,
	  libnice are enabled by default.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1788
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9637>

2025-09-02 07:16:27 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  device-monitor: Fix CMD vs PowerShell detection
	  PSModulePath is set in both powershell and cmd tabs spawned inside
	  Windows Terminal, and ComSpec is also set in both. PROMPT is the only
	  variable that is only set by cmd, which means it will also exist in a
	  powershell prompt spawned inside cmd, but in that case it's much harder
	  to figure out the parent shell. We'd have to look at the process tree.
	  This should be good enough for now.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9634>

2025-09-02 07:08:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  device-monitor: Double-escape \ to deal with gst_value_deserialize()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9634>

2025-09-01 16:39:02 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  device-monitor: Fix unnecessary quoting when serializing
	  Not all serialized values need quoting, for example booleans.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9634>

2025-09-01 16:26:21 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  device-monitor: Add quoting for powershell and cmd
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4620
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9634>

2025-08-19 09:15:03 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  rtpbasedepayload: Avoid potential use-after free
	  Clear the pointer after freeing the reference.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9581>

2025-05-15 18:33:30 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Update stream tags
	  parsebin does this, so should decodebin3.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9566>

2025-08-11 10:08:40 +0300  Andrey Khamukhin <andrew.khamukhin@hotmail.com>

	* gst/videorate/gstvideorate.c:
	* tests/check/elements/videorate.c:
	  videorate: don't hold the reference to the buffer in drop-only mode
	  Pushing the buffer via gst_pad_push () in transform_ip () function
	  causes downstream elements to process the buffer with a reference
	  count > 1. This leads to performance issue if there are downstream
	  elements which modify the buffer memory.
	  However, in drop-only mode this reference is not required.
	  So, let GstBaseTransform push the buffer in drop-only mode.
	  Fixes #4258
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9548>

2025-08-07 19:15:29 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.5

=== release 1.26.5 ===

2025-08-07 19:06:46 +0100  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.5

2025-08-06 15:27:32 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Add missing locking and NULL checks when adding URIs to messages
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4559
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9501>

2025-08-03 16:31:32 -0400  Xavier Claessens <xclaessens@netflix.com>

	* gst/audioconvert/gstaudioconvert.c:
	* tests/check/elements/audioconvert.c:
	  audioconvert: Fix regression when using a mix matrix
	  This fixes regression introduced by commit da3a1011. When a mix matrix
	  is set, we still want to set the default channel-mask on output caps.
	  Fixes: #4579
	  Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9499>

2025-08-05 12:44:38 +0100  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  device-monitor: Also accept utf8 in launch lines
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9498>

2025-08-05 12:40:21 +0100  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  device-monitor: Fix criticals when dumping non-string values
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9498>

2025-07-31 20:54:20 +1000  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Don't error out for unknown streams in default selection
	  If there is only unknown stream-type streams in the current collection
	  don't post an error straight away. This fixes a problem with RTSP
	  cameras and legacy upstream collection building, if the first
	  stream that rtspsrc outputs is the ONVIF metadata track. That
	  happens often on bandwidth-constrained camera inputs, as the
	  video and audio will naturally take longer to arrive.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9488>

2025-07-31 12:50:18 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  gst-device-monitor: Add shell quoting for launch lines
	  Launch-lines will be pasted into a shell, and `gst_value_serialize()`
	  yields strings that will likely be interpreted by the shell. For
	  example:
	  `gst-launch-1.0 ... ! osxaudiosink unique-id="AppleUSBAudioEngine:BEHRINGER:UMC202HD\ 192k:12345678:1\,2"`
	  The shell will remove the double-quotes `"` but keep the `\ ` which
	  means the output of `gst_value_deserialize()` will not be the original
	  string, and the launch line will not work.
	  So let's use `gst_value_serialize()` only if the string is non-ASCII,
	  and if it's ASCII and needs quoting, we do some shell quoting.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9486>

2025-07-31 12:49:32 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  gst-device-monitor: Don't loop unnecessarily when printing properties
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9486>

2025-07-31 20:05:15 +0900  Seungha Yang <seungha@centricular.com>

	* tools/gst-device-monitor.c:
	  device-monitor: Use gst_print instead of g_print
	  Avoid broken stdout output on Windows. Same change was made for
	  gst-launch in commit 493a3261a9757b5ade7aec289eb07221966f9eed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9470>

2025-07-22 15:23:30 +0200  Jaslo Ziska <jaslo@ziska.de>

	* ext/gl/gstgloverlay.c:
	  gloverlay: Recompute geometry when caps change
	  Set geometry_changed when setting caps so that the geometry is
	  recomputed correctly with the new dimensions.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9462>

2025-07-22 15:21:23 +0200  Jaslo Ziska <jaslo@ziska.de>

	* ext/gl/gstgloverlay.c:
	  gloverlay: Load texture after stopping and starting again
	  Set location_has_changed when stopping so that the texture will be
	  loaded when starting again.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9462>

2025-03-04 01:29:27 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst/videorate/gstvideorate.c:
	  videorate: add support for JPEG-XS
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9421>

2025-07-16 15:33:19 +0200  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.4

=== release 1.26.4 ===

2025-07-16 15:26:21 +0200  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.4

2025-07-08 17:06:25 +0100  Tim-Philipp Müller <tim@centricular.com>

	* po/hr.po:
	* po/ka.po:
	  gst-plugins-base: update translations

2025-06-11 01:45:43 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/alsa/gstalsasink.c:
	* ext/alsa/gstalsasrc.c:
	  alsa: free conf cache under valgrind
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9362>

2025-05-30 15:11:31 -0400  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/videoscale.c:
	  videoscale: Fix test for allowed caps
	  videoscale_get_allowed_caps_for_method() could leave holes in the
	  returned array, causing the test to skip some caps and not free them.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9347>

2025-05-30 15:08:32 -0400  Doug Nazar <nazard@nazar.ca>

	* gst-libs/gst/gl/gstglviewconvert.c:
	  glviewconvert: Fix a memory leak
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9347>

2025-05-30 15:07:07 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/gl/gstglstereosplit.c:
	  glstereosplit: Fix a leaked event
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9347>

2025-07-08 02:33:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  gst-device-monitor: Fix caps filter splitting
	  max_tokens=-1 means we will split on `:` in the caps as well, for
	  example caps features, and then discard those tokens.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9346>

2025-05-28 08:36:28 -0400  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/appsrc.c:
	  tests: appsrc: fix race accessing expected list
	  Without synchronization, a thread may still see an old value.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9343>

2025-05-28 08:42:15 -0400  Doug Nazar <nazard@nazar.ca>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: free fullname on failure
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9343>

2025-06-22 03:13:00 -0400  Doug Nazar <nazard@nazar.ca>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Print correct frame variable
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9314>

2025-06-30 09:26:29 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gststreamsynchronizer.c:
	* tests/check/elements/streamsynchronizer.c:
	  Revert "streamsynchronizer: Consider streams having received stream-start as waiting"
	  This reverts commit a1a189c07cb66af06d7047c74f6421bd36e3d66c.
	  It breaks the uriplaylistbin tests and needs further investigation.
	  See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4506
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9312>

2025-06-26 21:29:34 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.3

=== release 1.26.3 ===

2025-06-26 21:25:24 +0100  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.3

2025-06-12 16:32:51 -0400  Xavier Claessens <xclaessens@netflix.com>

	* gst/audioconvert/gstaudioconvert.c:
	* tests/check/elements/audioconvert.c:
	  audioconvert: Fix setting mix-matrix when input caps changes
	  When the number of input channels changes, application might have to set
	  a new mix-matrix. Application must set the new matrix before
	  audioconvert receives updated caps, otherwise negotiation would fail.
	  That means it should be allowed to set an invalid mix-matrix until we
	  receive new caps or next buffer.
	  This fixes a regression in GStreamer >=1.24.9 caused by:
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7363
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9290>

2025-06-13 09:54:33 -0400  Xavier Claessens <xclaessens@netflix.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: Replace g_warning with GST_WARNING_OBJECT
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9290>

2025-06-05 17:03:12 +0200  Thibault Saunier <tsaunier@igalia.com>

	* gst/playback/gstplaysink.c:
	* gst/playback/gststreamsynchronizer.c:
	  playsink: Fix race condition in stream synchronizer pad cleanup during state changes
	  Prevent race condition where gst_play_sink_do_reconfigure() could be called
	  from a pad probe while stream synchronizer pads are being released during
	  GST_STATE_CHANGE_PAUSED_TO_READY transition.
	  The race occurred when:
	  1. State change starts releasing stream synchronizer pads
	  2. Pads are unblocked earlier in the state change, allowing events to flow
	  3. A streaming thread triggers sinkpad_blocked_cb -> gst_play_sink_do_reconfigure
	  4. Reconfiguration tries to use already-released pad pointers
	  5. New pad creation fails with assertion in gst_pad_iterate_internal_links
	  The fix adds GST_PLAY_SINK_LOCK around the pad cleanup to ensure atomic
	  cleanup and prevent concurrent access during state transitions.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9263>

2025-06-18 19:03:17 +0200  Loïc Le Page <llepage@igalia.com>

	* gst-libs/gst/audio/gstaudioaggregator.c:
	  GstAudioAggregator: fix structure unref in peek_next_sample
	  The GstStructure attached to the audio sample in peek_next_sample() was
	  freed prematurely before usage as gst_sample_new() is taking full
	  ownership on it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9260>

2025-01-31 17:56:44 +0100  Théo Maillart <tmaillart@freebox.fr>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: never manually store stream-start
	  The copy of the exact same stream-start event prevents the multiqueue's sink
	  event function from being called because it is already stored on both pads at
	  link time
	  The text streams are no longer considered sparse by the multiqueue, so
	  interleave calculation is broken and makes us consume a lot of ram and we can
	  end up killed by the kernel because of this
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9250>

2025-06-07 13:32:18 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/gl/gstglutils.c:
	* ext/gl/gstglutils.h:
	* ext/gl/gstopengl.c:
	  gl: simplify private gst_gl_gst_meta_api_type_tags_contain_only()
	  Follow-up to backport commit 7dd465ba.
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9041#note_2923013
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9195>

2025-05-30 12:41:24 -0400  Xavier Claessens <xclaessens@netflix.com>

	* gst-libs/gst/app/gstappsink.c:
	  python: Fix pulling events from appsink
	  appsink.pull_object() is introspectable, but it needs a way to convert
	  the GstMiniObject to its GstEvent/GstSample subclass.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9178>

2025-06-04 11:57:38 +0200  Philippe Normand <philn@igalia.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebasebin: GstPad and GstStructure leak fixes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9173>

2025-06-04 12:41:35 +0200  Philippe Normand <philn@igalia.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebasebin: Make profile ownership explicit in StreamGroup
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9173>

2025-06-04 12:40:54 +0200  Philippe Normand <philn@igalia.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebasebin: Encoding profile ownership fixes
	  The profile argument passed to gst_encode_base_bin_set_profile is now
	  transfer-full. This issue was noticed after commit
	  6beb709d43d2023e7e5dc8f1ee1323bc28c9d1d8 which fixed profile refcount handling
	  in transcodebin.
	  Driving-by, an encoding profile leak was also fixed in _set_profile, in case
	  it's called for an already active element.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9173>

2025-05-30 09:40:21 +0200  Thibault Saunier <tsaunier@igalia.com>

	* sys/xvimage/xvimagesink.c:
	  doc: Add some explanation about the logic of when to post navigation message in code
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9169>

2024-11-14 21:36:19 +0100  Enrique Ocaña González <eocanha@igalia.com>

	* gst/playback/gststreamsynchronizer.c:
	* tests/check/elements/streamsynchronizer.c:
	  streamsynchronizer: Consider streams having received stream-start as waiting
	  When using the custom WebKitMediaSrc element (used by WebKit and able to
	  perform an initial seek in playbin), a stall caused by streamsynchronizer
	  was detected during an initial seek. The flow of events revealed that the
	  intertwining of the initial configuration of the streams with the reset
	  caused by the flush events from the seek left streamsynchronizer in an
	  inconsistent state:
	  streamsynchronizer0:sink_0 (video) events, starting before the seek:
	  stream-start --> Sets the stream to wait
	  flush-stop --> Clears the stream wait flag
	  caps
	  tag
	  segment
	  stream-collection
	  (buffers start to come and flow properly)
	  streamsynchronizer0:sink_1 (audio) events, happening after seek:
	  (no flush events, because the stream hadn't been initialized when the seek happened)
	  stream-start --> Sets the stream to wait
	  caps
	  segment
	  (stalled because the stream is in wait mode!)
	  The code in streamsynchronizer expects that all the streams are in wait
	  state before releasing all of them at once. The flush on the video stream
	  broke that assumption and that's why the audio stream is never released in
	  that scenario.
	  Avoiding the clearing of the wait flag on flush-stop isn't an actual solution
	  to the problem, as it creates other side effects and at least makes the
	  gst-editing-services/seek_with_stop test to timeout. The alternate solution
	  implemented in this patch consists on analyzing if the other streams different
	  from the one newly added (after the flush) aren't waiting (which would mean
	  that they've all been unlocked after all of them were waiting before), and,
	  in that case, mark the new stream as also not waiting.
	  A new test_stream_start_wait test case has been added to demonstrate this
	  problem. The test case creates a video stream, pushes a buffer, then
	  simulates a seek by pushing flush-start, flush-stop, stream-start and segment
	  events. Note that the flush-stop clears the video stream waiting flag.
	  After that, a new audio stream is created and stream-start and new segment
	  events are sent. Note that stream-start will set the audio stream to wait.
	  Then a buffer is pushed on each stream. In the failing case, the test hangs.
	  In the working case (after this fix), the test runs properly because the
	  fact of having seen a stream-start also helps to clear the wait flag.
	  A second new test_stream_start_wait_sparse test has also been added to prove
	  that this mechanism can also work with sparse streams (a special case of the
	  current stream-start handling code). This test behaves like the previous one,
	  but there's no video buffer after the seek (it'll come in the future, as the
	  stream is sparse, but actually never comes). The buffer after the seek in the
	  audio stream starts at its due time. Streamsynchronizer is able to ignore
	  the wait for the video stream and produce audio buffers on time.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9160>

2025-05-30 00:03:12 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.2

=== release 1.26.2 ===

2025-05-29 23:56:43 +0100  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.2

2025-05-08 12:46:40 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/subparse/gstsubparse.c:
	  subparse: Make sure that subrip time string is not too long before zero-padding
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4419
	  Fixes CVE-2025-47806
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9135>

2025-05-08 09:14:15 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/subparse/gstsubparse.c:
	  subparse: Check for valid UTF-8 before cleaning up lines and check for regex replace errors
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4418
	  Fixes CVE-2025-47807
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9135>

2025-05-08 09:04:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/subparse/tmplayerparse.c:
	  tmplayer: Don't append NULL + 1 to the string buffer when parsing lines without text
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4417
	  Fixes CVE-2025-47808
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9135>

2025-05-29 00:14:50 +0300  Vasiliy Doylov <nekocwd@mainlining.org>

	* ext/gl/gstglfiltershader.c:
	  glshader: recompile shader on pipeline restart
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9125>

2025-05-26 19:00:36 +0200  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff-media: fix MS and DVI ADPCM av_bps calculations
	  Align the calculations for the number of samples per block with the
	  calculations in adpcmdec.
	  For MS ADPCM we have in adpcmdec:
	  samples = (blocksize - 7 * dec->channels) * 2 + 2 * dec->channels;
	  outsize = 2 * samples;
	  outbuf = gst_buffer_new_and_alloc (outsize);
	  This gives us the total output byte size in 16 bits samples. To get back
	  to the samples, dividing by the channels and 2, we get the right samples per
	  block as:
	  int spb = ((strf->blockalign / strf->channels) - 7) * 2 + 2;
	  Which we can then use to calculate the bitrate in riff-media.
	  A similar calculation for DVI ADPCM is needed to get the right bitrate
	  in all cases.
	  Tested with the sample in https://bugzilla.gnome.org/show_bug.cgi?id=636245
	  and another (failing before this patch) sample.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9121>

2025-05-27 19:20:35 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/pango/gstbasetextoverlay.c:
	  textoverlay: fix shading for RGBx/RGBA pixel format variants
	  ... for cases where there's padding at the end of each row.
	  Fixes #4414.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9111>

2025-05-21 10:01:24 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/playback/gstdecodebin3.c:
	  gstreamer: A few small memory cleanups
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9101>

2025-05-21 10:04:59 -0400  Doug Nazar <nazard@nazar.ca>

	* gst-libs/gst/gl/egl/gstglcontext_egl.c:
	  glcontext/egl: Free dma_formats if someone else already initialized
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9101>

2025-05-14 13:54:55 -0400  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/opus.c:
	* tests/check/meson.build:
	  tests: opus: Update channel support and add to meson
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9090>

2025-05-14 14:42:19 -0400  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/audioresample.c:
	  tests: Switch to GST_AUDIO_NE()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9089>

2025-05-14 14:39:17 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/volume/gstvolume.c:
	* tests/check/elements/volume.c:
	  volume: Switch to GST_AUDIO_NE()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9089>

2025-05-14 14:38:41 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/audiomixer/gstaudiointerleave.c:
	* gst/audiomixer/gstaudiomixer.c:
	  audiomixer: Switch to GST_AUDIO_NE()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9089>

2025-05-14 14:37:47 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/adder/gstadder.c:
	* tests/check/elements/adder.c:
	  adder: Switch to GST_AUDIO_NE()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9089>

2025-05-13 19:42:37 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/gl/gstglfiltershader.c:
	  gstglfiltershader: Free various props before set & during cleanup
	  gst_object_replace() takes a reference so no need to dup object.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9088>

2025-05-21 20:29:06 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* meson_options.txt:
	  meson: Add a monorepo-wide qt-method option and yield to it
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9063>

2025-05-21 20:25:26 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	  meson: Rework build files for qt examples in -base
	  The build files had quite a few things wrong:
	  * Not using the method: kwarg, which can cause the wrong Qt to be
	  used for building
	  * There was no way to enable the build for them
	  * Qt was being detected multiple times, differently
	  * Unnecessary check for libGL
	  * have_cxx was being used incorrectly
	  * Qt tool detection was outdated
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9063>

2025-05-17 11:40:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/gl/gstglcolorbalance.c:
	* ext/gl/gstglcolorconvertelement.c:
	* ext/gl/gstglcolorscale.c:
	* ext/gl/gstgldownloadelement.c:
	* ext/gl/gstgluploadelement.c:
	* ext/gl/gstglutils.c:
	* ext/gl/gstglutils.h:
	* ext/gl/gstopengl.c:
	  gl: Implement basetransform meta transform function
	  This makes sure we can pass through more metas correctly, e.g.
	  GstVideoOverlayComposition meta.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4422
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9041>

2025-05-18 11:30:24 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/videoconvertscale/gstvideoconvertscale.c:
	  videoconvertscale: Use new gst_meta_api_type_tags_contain_only() API
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9041>

2025-05-20 08:21:51 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Don't hold play items lock while releasing pads
	  Releasing the pad can cause messages that call back into the message
	  handler of uridecodebin3 and take exactly the same lock again.
	  Fix #4443
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9047>

2025-05-18 12:00:55 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: Remove 0.10 hardware caps handling
	  This also reverts c02d41c2. videoconvert and videoscale are supposed to support
	  raw video with any caps features as long as no conversion is actually necessary,
	  and assuming they don't breaks usage of GstVideoOverlayCompositionMeta with e.g.
	  dmabuf or GL memory caps.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4353
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9045>

2025-05-13 08:20:30 -0400  Doug Nazar <nazard@nazar.ca>

	* tests/check/elements/audiomixer.c:
	  audiomixer: Change test to use native endian audio format
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9025>

2025-05-13 08:19:54 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/videoconvertscale/gstvideoconvertscale.c:
	  videoconvertscale: Use correct variable size for gst_structure_get()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9025>

2025-04-13 19:31:23 +0200  Adrian Perez de Castro <aperez@igalia.com>

	* ext/alsa/gstalsa.c:
	  alsa: Avoid infinite loop in DSD rate detection
	  Stop testing DSD rates in gst_alsa_detect_dsd_rates() if the rate becomes zero
	  or negative. This avoids an infinite loop if gst_alsa_probe_supported_formats()
	  is used on a PCM sink defined like the following in the ALSA configuration file:
	  pcm.buggy {
	  type plug
	  slave.pcm "buggy_volume"
	  hint.description "Causes an infinite loop in GStreamer"
	  }
	  pcm.buggy_volume {
	  type softvol
	  slave.pcm "buggy_dmix"
	  control.name "buggy_volume"
	  }
	  pcm.buggy_dmix {
	  type dmix
	  ipc_key 12345
	  slave {
	  pcm "hw:0,0"
	  period_size 1024
	  buffer_size 4096
	  }
	  }
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9005>

2025-04-26 15:04:01 +0200  Robert Mader <robert.mader@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: Only add texture-target field to GL caps
	  So far we simply ignored it for MEMORY_DMABUF passthrough caps
	  without known negative cosequences, but with upcoming more complicated
	  caps negotiations it's becoming an issue, thus fix it.
	  Fixes: 7e71d4f753 ("gl: upload: Add DMA_DRM passthrough upload")
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8998>

2025-05-02 08:56:19 +0200  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* ext/pango/gstbasetextoverlay.c:
	* gst-libs/gst/allocators/gstdrmdumb.c:
	* gst-libs/gst/rtp/gstrtpbasedepayload.c:
	  properties: add G_PARAM_STATIC_STRINGS where missing
	  "Hold on, I know you need to generate the registry, but let me just
	  create copies of all those strings first", Framework whispered
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8962>

2025-05-01 11:48:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/sdp/gstmikey.c:
	  mikey: Avoid infinite loop while parsing MIKEY payload with unhandled payload types
	  Skip over the unhandled payload types for now, and error out if an unknown
	  payload type is found.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3314
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8947>

2025-04-26 13:38:06 +0000  Biswapriyo Nath <nathbappai@gmail.com>

	* gst-libs/gst/audio/gstaudioutilsprivate.c:
	  gstaudioutilsprivate: Fix gcc 15 compiler error with function pointer
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8910>

2025-04-01 17:58:14 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/vorbis/meson.build:
	  meson: Add include_type: 'system' everywhere to squelch wrap warnings
	  Wrap dependencies add a ton of warnings with the latest GCC in Fedora
	  42. Squelch them by specifying that these dependencies are not
	  a part of the gstreamer project, and should be treated as system deps.
	  libsoup needs some porting work for the bump, and vorbis/lame are
	  already at their latest releases.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8919>

2025-04-24 20:26:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.1
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8890>

=== release 1.26.1 ===

2025-04-24 20:20:14 +0100  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.1

2025-04-15 14:24:12 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideoencoder.c:
	  videoencoder: Use the correct segment and buffer timestamp in the chain function
	  The only thing that can be used in the chain function is the input segment. The
	  output segment might not be available at all yet or out of sync with the current
	  input segment.
	  Also because of that, the unadjusted timestamp has to be used for the
	  calculations as the adjustment is only part of the output segment.
	  This fixes the deadline calculation and the handling of force-keyunit events for
	  encoders using frame reordering (i.e. setting a minimum PTS).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8843>

2025-04-07 15:07:56 +0100  Philippe Normand <philn@igalia.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: Don't push new packets if there is a pending seek
	  There was a race condition where the demuxer would seek back to beginning after
	  determining the duration and while that seek was in progress one pad would
	  attempt to push a new buffer downstream, leading to a critical warning in
	  gst_pad_push().
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8834>

2025-04-10 16:19:42 +0900  Hou Qi <qi.hou@nxp.com>

	* ext/gl/gstgldownloadelement.c:
	  gldownload: unref glcontext after usage
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8813>

2025-03-11 09:30:29 +0100  Carlos Rafael Giani <crg7475@mailbox.org>

	* gst-libs/gst/audio/gstaudiobasesink.c:
	  audiobasesink: Fix custom slaving driftsamples calculation
	  driftsamples currently uses the requested skew directly, even if it
	  exceeds cexternal.
	  Use the approach that skew_slaving uses to fix this. As a side benefit,
	  this makes the custom_slaving and skew_slaving code easier to compare.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8811>

2025-03-09 22:00:06 +0100  Carlos Rafael Giani <crg7475@mailbox.org>

	* tests/examples/audio/audiobasesink-custom-clock-slaving.c:
	* tests/examples/audio/meson.build:
	  examples: Add custom audio clock slaving callback example
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8811>

2025-04-03 13:20:50 +1100  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: fix YUVA<->RGBA conversions
	  Alpha should not impact the YUV/RGB conversion and should simply be copied
	  over.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4339
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8769>

2025-03-31 12:18:42 +0200  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* gst-libs/gst/audio/gstaudioaggregator.c:
	  audioaggregator: fix chaining up to parent class (again)
	  An error was added in !8416, it was calling to the
	  wrong parent class
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8743>

2025-03-14 22:10:56 -0400  Doug Nazar <nazard@nazar.ca>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Free various props before being set
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8714>

2025-03-14 22:10:10 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/theora/gsttheoraenc.c:
	  theoraenc: Free various props before being set
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8714>

2025-03-14 22:09:21 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/gl/gstglbumper.c:
	* ext/gl/gstgldifferencematte.c:
	* ext/gl/gstgloverlay.c:
	  gl: Free various props during cleanup
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8714>

2025-03-14 22:07:04 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/alsa/gstalsamidisrc.c:
	  alsamidisrc: free ports during finalize()
	  If the element is never start/stopped the ports variable will leak.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8714>

2025-03-14 19:14:43 -0400  Doug Nazar <nazard@nazar.ca>

	* ext/alsa/gstalsadeviceprovider.c:
	  all: Annotate *_set_property() contructor only props without free
	  Properties that are marked constructor only aren't required to be freed
	  before g_value_dup_string() as they can only be called once during construction.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8714>

2025-03-24 15:56:01 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst-libs/gst/gl/egl/gsteglimage.c:
	  gl: eglimage: warn the reason of export failure
	  So people debugging could know what's happening at debugging.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8713>

2025-03-24 16:53:46 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst-libs/gst/gl/egl/gsteglimage.c:
	  gl: eglExportDMABUFImageQueryMESA expects modifiers to be an array
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8713>

2025-03-15 12:07:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideotimecode.c:
	* tests/check/libs/videotimecode.c:
	  videotimecode: Add missing 119.88fps support to some functions
	  And while at it generalize the drop frame handling to all integer multiples
	  of 30000/1001 fps.
	  Also adjust tests accordingly and add some other missing test.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8704>

2025-03-15 11:00:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideotimecode.c:
	* tests/check/libs/videotimecode.c:
	  videotimecode: Fix conversion of timecode to datetime with drop-frame timecodes
	  gst_video_time_code_to_date_time() simply calculated the date time based on
	  adding the hours/minutes/seconds to the daily jam. This causes a gap every full
	  minute (except for every 10th minute) with drop-frame timecodes as the first 2
	  (29.97fps) or 4 (59.94fps) timecodes are skipped (not frames!), e.g. with
	  29.97fps:
	  timecode: 12:00:59;28  12:00:59;29  12:01:00;02  12:01:00;03
	  time    : 12:00:59.950 12:00:59.983 12:01:00.017 12:01:00.050
	  and not
	  time    : 12:00:59.934 12:00:59.968 12:01:00.067 12:01:00.100
	  |-- gap of 2 frames --|
	  The correct calculation would be to use gst_video_time_code_nsec_since_daily_jam()
	  and add that to the daily jam.
	  Also add a test for this.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8704>

2025-01-22 15:02:03 +0100  Marc Leeman <marc.leeman@gmail.com>

	* gst-libs/gst/audio/meson.build:
	* meson.build:
	  meson.build: test for and link against libatomic if it exists
	  It's needed on some platforms for some subset (or all) atomic operations and
	  checking for the cases when it's actually needed is quite complex.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4300
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8647>

2025-03-14 09:37:16 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Don't avoid `parsebin` even if we have a matching decoder
	  This is too brittle, there is no guarantee that the input stream has been
	  properly parsed.
	  There is another check above (is_input_parsed) that will skip that if the
	  content came from `urisourcebin` and had a parser applied
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4308
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8643>

2025-03-12 10:30:04 +0100  Olivier Blin <olivier.blin@softathome.com>

	* ext/alsa/gstalsadeviceprovider.c:
	  alsadeviceprovider: Fix leak of Alsa longname
	  Detected by ASan.
	  As a drive-by fix, use free() instead of g_free() in gstalsadeviceprovider.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8632>

2025-02-18 15:53:46 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Make parsebin activation more reliable
	  `parsebin` is potentially added by a `typefind` callback.
	  That `typefind` was activated by a `READY_TO_PAUSED` state change on `urisourcebin`
	  We want to ensure that it is the "setup_parsebin_for_slot" method that activates
	  the underlying `parsebin`, and not the external state-change.
	  Otherwise we would risk a potential deadlock where elements activating in
	  `parsebin`, and which would cause the upstream `typefind` to switch scheduling
	  mode, would not be able to acquire the STREAM_LOCK of the `typefind` task.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4225
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8622>

2025-03-11 20:23:16 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.26.0
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8618>

=== release 1.26.0 ===

2025-03-11 20:14:44 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* README.md:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.26.0

2025-03-11 12:43:55 +0100  Alyssa Ross <hi@alyssa.is>

	* gst/tcp/gsttcpsrcstats.c:
	  tcpclientsrc, tcpserversrc: fix building for musl
	  musl only exposes struct tcp_info when _GNU_SOURCE is defined.
	  Fixes: b4bef7fd35 ("tcpclientsrc, tcpserversrc: fix tcp stats gathering")
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8612>

2025-02-28 14:24:09 +0900  Dongyun Seo <dongyun.seo@lge.com>

	* gst/playback/gstplaybin3.c:
	  playbin3: fix unlocking twice
	  fix unlocking playbin3 lock twice.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8570>

2025-02-25 12:52:19 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/gl/gstglupload.c:
	  Revert "glupload: Add formats supported by #GstGLMemory to raw caps when generating sink pad caps"
	  This reverts commit cf30e875de7b8f04730753fb35869d0f7338b6e5.
	  This broke caps negotiation on Wayland/EGL with:
	  videotestsrc ! glupload ! glimagesinkelement
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4240
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8552>

2025-02-23 23:52:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.25.90

=== release 1.25.90 ===

2025-02-23 23:44:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.25.90

2025-02-23 16:55:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/ka.po:
	* po/lt.po:
	* po/lv.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  gst-plugins-base: update translations

2025-02-18 16:36:28 +1100  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/egl/gstgldisplay_egl.c:
	* gst-libs/gst/gl/egl/gstgldisplay_egl.h:
	  gldisplay/egl: Add API for overriding foreign-ness of the EGLDisplay
	  Scenario is using wayland with this pipeline: videotestsrc ! glupload ! qml6glsink.
	  First pipeline and qml construction works just fine.  However if all GStreamer
	  and QML scenegraph resources are removed, GStreamer will call eglTerminate() and
	  sever the connection to the display server for Qt.  When Qt attempts to do any
	  further GL operations like construct a new QML scene, it can crash in any number
	  of places as libEGL will start returning NULL or other unexpected values.
	  What we really need is to ensure that if an external API (e.g. Qt) will
	  eventually call eglTerminate(), there is no need for GStreamer to call
	  eglTerminate().  This is what the foreign display flags allows setting.
	  There is also another possible scenario where one may like to make GStreamer
	  assume ownership of an EGLDisplay and thus eventually call `eglTerminate()`.  As
	  such, it is now also possible to mark an GstGLDisplayEGL as non-foreign.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8504>

2025-02-19 09:04:07 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	  video: dma-drm: Fix MT2110R/T DRM mapping
	  The base format for these should be NV15 (10bit) rather then NV12.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8524>

2025-02-20 19:57:48 +1100  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/video/convertframe.c:
	  convertframe: Fix video crop meta handling
	  Cropping parameters were being miscalculated - getting
	  the output width/height wrong when an x/y crop offset
	  was given.
	  Cropping was also incorrectly being applied twice (because at
	  some point after the convertframe code was written,
	  `videocrop` also started paying attention to the
	  GstVideoCropMeta, but not in useful ways for this purpose). Add
	  a buffer probe to strip the crop meta from the input buffer
	  so videocrop can do its job correctly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8526>

2025-02-19 09:32:43 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst-libs/gst/gl/gstglupload.c:
	  gl: upload: use empty caps if transform_caps returns NULL
	  transform_caps() from the upload methods can return NULL in some cases, so fall
	  back to empty caps in that case.
	  This can happen if "Raw Data" is the currently selected method and new caps with
	  memory:DMABuf caps feature are negotiated.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8514>

2025-02-14 13:04:27 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/audiorate/gstaudiorate.c:
	  audiorate: Take the tolerance into account when filling gaps
	  The property is defined as:
	  > The difference between incoming timestamp and next timestamp must exceed
	  > the given value for audiorate to add or drop samples.
	  so if the gap duration < tolerance, we should not act.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8498>

2025-02-19 02:08:06 +0000  Markus Ebner <info@ebner-markus.de>

	* gst-libs/gst/tag/gstvorbistag.c:
	  gstvorbistag: Add support for lyrics tag
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8513>

2024-12-05 14:05:52 +0100  Benjamin Gaignard <benjamin.gaignard@collabora.com>

	* gst/typefind/gsttypefindfunctions.c:
	* gst/typefind/gsttypefindfunctionsplugin.c:
	* gst/typefind/gsttypefindfunctionsplugin.h:
	  typefind: Add typefinder for AV1 obu/annexb raw streams
	  AV1 specification [1] define 2 AV1 raw bitstreams storage
	  formats without containers:
	  - OBU in chapiter 5.2
	  - Annexb in chapiter 11.
	  Implement a detection function for the both cases mostly
	  by testing OBU forbidden, type, has_size_field and reserved bits.
	  For annexb case testing if temporal unit size, frame unit size and
	  obu length are valid. If they are check that the first OBU is
	  a temporal delimiter.
	  [1] https://aomediacodec.github.io/av1-spec/av1-spec.pdf
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8502>

2024-12-27 13:31:41 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Don't hold play items lock while activating source items
	  Activating them can cause messages that call back into the message handler of
	  uridecodebin3 and take exactly the same lock again.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8215>

2020-09-25 21:06:44 +0900  Seungha Yang <seungha@centricular.com>

	* gst-libs/gst/gl/gstglupload.c:
	* tests/check/libs/gstglupload.c:
	  glupload: Fix for wrongly recognized reconfigure condition
	  gst_gl_upload_transform_caps() method might return non-fixed
	  caps (texture-target for example) but priv->out_caps is fixed one
	  so the former (non-fixed caps) is superset.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8492>

2024-10-24 16:50:49 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: descriptions: add Hap video codec
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7717>

2025-02-10 16:47:16 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: Don't skip all other methods than the currently selected one when transforming caps
	  This leads to spurious negotiation failures because the configured method can
	  change over time and caps queries (and thus transform_caps) are happening
	  independently from configuring caps. Instead prefer the transformed caps of the
	  current method, but always also return the transformed caps for all other
	  methods. Previously all other methods would've only been used if the current
	  method returned empty caps. If a different method is needed later when
	  configuring the caps, it will be and was selected regardless.
	  Later during caps fixation, prefer the caps of the current method too for the
	  fixated caps if possible in any way.
	  This should preserve the desired behaviour of preferring the current method if
	  possible but to change to a different method if nothing else is possible, while
	  also returning consistent (and not too narrow) caps every time.
	  The way how the current method was checked was also racy as the current method
	  might change at any moment during caps query handling, and apart from
	  inconsistent results also a NULL pointer dereference was possible here. Use the
	  GST_OBJECT_LOCK to protect access to the current method like in other places.
	  This part of the code was introduced in f349cdccf5e1538f3eb9caa31458b53fdd81671b
	  and tried to be fixed multiple times over the years without addressing the root
	  cause of caps queries and caps configuration happening independently from each
	  other, e.g. in !2687 and !2699.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8431>

2025-02-14 12:02:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/gl/gstgluploadelement.c:
	  gluploadelement: Fix typo in debug output
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8431>

2025-02-10 14:41:01 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video-overlay-composition.c:
	  video-overlay-composition: Fix meta scale transformation for xscale != yscale
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8431>

2025-02-07 11:34:18 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/videoconvertscale/gstvideoconvertscale.c:
	  videoconvertscale: Explicitly handle overlaycomposition meta caps feature
	  Otherwise it will be dropped unnecessarily. videoconvertscale can always pass it
	  through, no matter if it does conversion or not.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4161
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8431>

2025-01-09 10:03:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video-chroma.h:
	* gst-libs/gst/video/video-info.c:
	  video-info: Validate chroma-site when parsing caps and set defaults if none is set
	  Previously there was no validation at all and the defaults were set if the
	  colorimetry was not set or invalid, but there's not really any connection
	  between colorimetry and chroma-site.
	  More validation could make sense in the future.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8258>

2025-02-11 00:02:32 +0100  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* ext/gl/gstglvideomixer.c:
	  glvideomixer: fix missing GObject vtable chainups
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8452>

2025-02-06 11:39:02 +0100  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* tests/examples/gl/gtk/3dvideo/mviewwidget.c:
	  examples/gl/gtk: fix chaining up GObject's constructed virtual method
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8416>

2025-02-06 11:29:34 +0100  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* ext/gl/gstglvideoflip.c:
	* ext/gl/gstglvideomixer.c:
	  gl: fix chaining up GObject's constructed virtual method
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8416>

2025-02-06 00:20:06 +0100  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* gst-libs/gst/audio/gstaudioaggregator.c:
	  audioaggregator: fix chaining up GObject's constructed virtual method
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8416>

2025-02-07 12:33:27 +0100  Carlos Bentzen <cadubentzen@igalia.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	* gst-libs/gst/pbutils/codec-utils.h:
	* tests/check/libs/pbutils.c:
	  pbutils: add profile-tier-level functions for VVC/H.266
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5088>

2025-02-09 17:47:32 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.25.50

=== release 1.25.50 ===

2025-02-09 17:35:17 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.25.50

2025-02-05 15:28:35 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	  video: Add DRM for NV12_10LE40 to DRM_FORMAT_NV15
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8426>

2025-02-03 13:20:21 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	  video: dma-drm: Extend the format mapping
	  These formats were mapped in video4linux2 plugin, but not here. Add them
	  so te video4linux2 plugin can use the global mapping.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8404>

2025-02-03 15:31:47 +0100  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	* tests/check/libs/video.c:
	  video: Add support for big endian DRM formats
	  When a format is big endian, the 8bith of the fourcc is set. Handle this by
	  using a "_BE" suffix in serialization. The patch also update the design document
	  and introduce a unit test.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8404>

2019-04-11 11:24:34 +0100  Philippe Normand <philn@igalia.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: Fix racy decode error
	  Sometimes the seek to the end of file to determine the duration would trigger a
	  reset of the source pads, that would confuse the decoder downstream and trigger
	  an error. So the proposed fix is to not reset pads when the segment event being
	  processed is the consequence of a seek performed to determine the duration.
	  Fixes #4212
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/937>

2025-02-03 11:12:59 +0100  Michael Tretter <m.tretter@pengutronix.de>

	* gst/videotestsrc/meson.build:
	  videotestsrc: fix build if adder is disabled
	  The videotestsrc is not build if the adder plugin is disabled. This is a
	  copy/paste error introduced in Commit 3de86b2b9725 ("docs: port plugins to
	  explicit sources").
	  Fix the check to actually test the 'videotestsrc' option.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8399>

2024-06-11 17:46:11 +0200  Ruben Gonzalez <rgonzalez@fluendo.com>

	* meson.build:
	  meson: use nls option to ENABLE_NLS
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7017>

2023-09-29 18:10:09 +0200  Tim-Philipp Müller <tim@centricular.com>

	* scripts/update-orc-dist-files.py:
	  scripts: update update-orc-dist-files.py scripts for new gst-indent
	  And fix python indentation with autopep8
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5192>

2024-11-14 08:40:19 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	  video: Add DMA_DRM mapping for NV12_16L32S / MT2110T / MT2110R
	  This is used on Meditek SoC and Mesa Panfrost [0] has enabled the 8bit variant
	  with a faster GL shader implementation then ours.
	  [0] https://gitlab.freedesktop.org/mesa/mesa/-/merge_requests/31899
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7892>

2025-01-28 13:48:16 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/ext/drm_fourcc.h:
	* gst-libs/gst/video/video-info-dma.c:
	  video: Import a verbatim copy of drm_fourcc.h
	  Imported from drm-tip [0] at commit 3ab334814dc7dff39075e055e12847d51878916e.
	  As the copied definition became bigger an bigger, it was quite apparent this
	  was more work then needed. Simply replace manual copies with a verbatim
	  official copy of it.
	  [0] https://gitlab.freedesktop.org/drm/tip/-/blob/drm-tip
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7892>

2025-01-13 16:19:26 +0100  Tomas Granath <tomas.granath@outlook.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: Do not change total duration until new duration is found
	  After calculating a correct duration the oggdemux in some cases sets the duration to GST_CLOCK_TIME_NONE.
	  After that any seek will fail due to the oggdemux calculating a target time after the actual duration.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8296>

2025-01-24 08:29:24 +0100  Jochen Henneberg <jochen@centricular.com>

	* gst/audiorate/gstaudiorate.c:
	* tests/check/elements/audiorate.c:
	  audiorate: Account for in buffer samples if fill samples are added
	  If fill buffers are added to compensate for missing samples the input
	  buffer is pushed as well and has to be added to output samples count.
	  This ensure that add - drop == out - in.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8358>

2025-01-12 12:44:29 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video-overlay-composition.c:
	  video-overlay-composition: Handle the SCALE meta transformation
	  Simply update the render rectangle according to the scale factor. As only the
	  render rectangle is changed, the seqnum of the rectangle stays intact and no
	  additional GPU uploads are necessary.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8288>

2025-01-12 12:43:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video-overlay-composition.c:
	  video-overlay-composition: Declare the video/size/orientation tags for the meta
	  If any of these change then the meta needs to be updated one way or another.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8288>

2025-01-20 13:23:50 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/meson.build:
	  docs: explicitly list gir files as depends for generating configs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8324>

2025-01-15 17:36:00 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/meson.build:
	* gst-libs/gst/allocators/meson.build:
	* gst-libs/gst/app/meson.build:
	* gst-libs/gst/audio/meson.build:
	* gst-libs/gst/gl/meson.build:
	* gst-libs/gst/pbutils/meson.build:
	* gst-libs/gst/riff/meson.build:
	* gst-libs/gst/rtp/meson.build:
	* gst-libs/gst/rtsp/meson.build:
	* gst-libs/gst/sdp/meson.build:
	* gst-libs/gst/tag/meson.build:
	* gst-libs/gst/video/meson.build:
	* gst-libs/meson.build:
	  docs: generate hotdoc configs for libraries with our helper script
	  With this patch, configure time is identical no matter whether doc is
	  enabled or not.
	  The configuration files also now contain explicitly-listed sources with
	  no wildcards.
	  For the four libraries where hotdoc needs to use clang to generate the
	  documentation (as opposed to the rest of the libraries where hotdoc uses
	  the gir), the script will call pkg-config to determine the appropriate
	  C flags.
	  This means a side effect of this patch is that pkg-config files are now
	  generated for the gstadaptivedemux and gstopencv libraries.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8312>

2025-01-17 16:51:22 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* meson.build:
	  meson: bump minimum version to 1.4 in every subprojects
	  36c01d05797ad9c7778939c54870f979bdcbba1f bumped to 1.4 for gst-devtools
	  and the root project, but we usually keep those in sync everywhere.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8312>

2025-01-07 18:55:02 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst/videoconvertscale/gstvideoconvertscale.c:
	  videoconvertscale: Handle large pixel aspect ratios
	  Some pixel aspect ratios found in dash streams have very large numerators and
	  denominators (while being close to 1:1). These values can cause integer overflow
	  during multiplication, leading to negotiation failures.
	  Add fallback path using gdouble when integer multiplication would overflow,
	  trading some precision for reliability instead of failing outright.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8262>

2025-01-14 15:00:43 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.25.1

=== release 1.25.1 ===

2025-01-14 14:52:48 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-base.doap:
	* meson.build:
	  Release 1.25.1

2025-01-13 18:10:31 +0100  Mathieu Duponchelle <mathieu@centricular.com>

	* docs/meson.build:
	* ext/alsa/meson.build:
	* ext/cdparanoia/meson.build:
	* ext/gl/meson.build:
	* ext/libvisual/meson.build:
	* ext/ogg/meson.build:
	* ext/opus/meson.build:
	* ext/pango/meson.build:
	* ext/theora/meson.build:
	* ext/vorbis/meson.build:
	* gst/adder/meson.build:
	* gst/app/meson.build:
	* gst/audioconvert/meson.build:
	* gst/audiomixer/meson.build:
	* gst/audiorate/meson.build:
	* gst/audioresample/meson.build:
	* gst/audiotestsrc/meson.build:
	* gst/compositor/meson.build:
	* gst/debugutils/meson.build:
	* gst/dsd/meson.build:
	* gst/encoding/meson.build:
	* gst/gio/meson.build:
	* gst/meson.build:
	* gst/overlaycomposition/meson.build:
	* gst/pbtypes/meson.build:
	* gst/playback/meson.build:
	* gst/rawparse/meson.build:
	* gst/subparse/meson.build:
	* gst/tcp/meson.build:
	* gst/typefind/meson.build:
	* gst/videoconvertscale/meson.build:
	* gst/videorate/meson.build:
	* gst/videotestsrc/meson.build:
	* gst/volume/meson.build:
	* meson.build:
	* sys/meson.build:
	* sys/ximage/meson.build:
	* sys/xvimage/meson.build:
	  docs: port plugins to explicit sources
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8273>

2025-01-04 15:37:03 +0100  Edward Hervey <edward@centricular.com>

	* ext/vorbis/gstvorbisdec.c:
	  vorbisdec: Fix check for "-1" bitrate values
	  The check is exactly the same, but more explicit.
	  Original commit that introduced the check is
	  20fb58be198e7d76ece4e6c635cda7c919fad6d2:
	  ---
	  vorbisdec: don't put invalid bitrate values into the taglist
	  Bitrates are stored as 32-bit signed integers in the vorbis
	  identification headers, but seem to be read incorrectly,
	  namely as unsigned 32-bit integers, into the vorbis structure
	  members which are of type long, which makes our check for
	  values <= 0 fail with files that put -1 in there for unset
	  values.
	  ---
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

2025-01-03 12:03:58 +0100  Edward Hervey <edward@centricular.com>

	* meson.build:
	  base: Enable more warning flags
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

2025-01-03 11:58:17 +0100  Edward Hervey <edward@centricular.com>

	* gst/compositor/blend.c:
	  compositor: Don't use CLAMP for unsigned values
	  Just use MIN, we already know it's above 0
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

2025-01-03 11:56:34 +0100  Edward Hervey <edward@centricular.com>

	* tools/gst-play.c:
	  gst-play: Move esc-handling to the right place
	  It's meant to fall through to the default handler of the switch (for
	  debugging). Some intermediary commits broke that.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

2025-01-10 11:05:06 +0100  Edward Hervey <edward@centricular.com>

	* ext/libvisual/meson.build:
	  libvisual: Fix for libvisual headers
	  They use old style definition
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

2025-01-03 10:36:17 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/tag/gstxmptag.c:
	* gst-libs/gst/video/gstvideopool.c:
	* tests/check/elements/encodebin.c:
	* tests/check/libs/rtp.c:
	  base: Do not use old-style definition
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

2025-01-03 10:35:57 +0100  Edward Hervey <edward@centricular.com>

	* ext/theora/gsttheoradec.c:
	* ext/theora/gsttheoraparse.c:
	* ext/vorbis/gstvorbisparse.c:
	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/audio/gstaudioencoder.c:
	* gst-libs/gst/pbutils/codec-utils.c:
	* gst-libs/gst/riff/riff-media.c:
	* gst-libs/gst/tag/gsttagdemux.c:
	* gst-libs/gst/tag/id3v2.c:
	* gst/playback/gstplaybin2.c:
	* gst/playback/gsturisourcebin.c:
	* gst/videorate/gstvideorate.c:
	* gst/videotestsrc/gstvideotestsrc.c:
	* tests/examples/playback/playback-test.c:
	  base: Fix fallthrough declarations
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

2025-01-03 12:01:37 +0100  Edward Hervey <edward@centricular.com>

	* gst/audiorate/gstaudiorate.c:
	  audiorate: Fix unsigned integer comparision
	  next_offset uses a magic "-1" value for unset values. Use that value and not
	  checking whether it's positive (it always is, it's unsigned)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8286>

2025-01-03 11:59:43 +0100  Edward Hervey <edward@centricular.com>

	* gst/videorate/gstvideorate.c:
	  videorate: Avoid overflow in calculations
	  base_ts and next_end_ts are both guint64. Use explicit check to avoid "negative"
	  timestamps (which would overflow, not be negative).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8286>

2025-01-03 11:59:05 +0100  Edward Hervey <edward@centricular.com>

	* tests/check/libs/rtspconnection.c:
	  test: Fix unsigned integer usage
	  num_queued is unsigned, it'll never be below 0.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8286>

2025-01-03 11:57:57 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Add missing break in switch
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8286>

2025-01-09 11:08:57 -0600  Olivier Crête <olivier.crete@collabora.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: Add missing nullable annotation to new APIs
	  Those APIs are documented to sometimes return NULL.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8271>

2024-12-27 20:35:44 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/video/video-info.c:
	  video-info: add "field-order" field to serialised caps also for interlace-mode=fields
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8219>

2024-12-27 20:33:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/video/gstvideoencoder.c:
	  videoencoder: copy over the field order for more interlace modes
	  .. when copying over video info things from the input state
	  to the output state.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8219>

2024-10-25 17:00:48 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst-libs/gst/app/gstappsink.c:
	  appsink: fix initial pile-up of caps events
	  appsink was queuing the caps event both in its GstBaseSinkClass.set_caps()
	  and its GstBaseSinkClass.sink_event() implementation, but only signaling
	  new_event from `sink_event`.
	  This caused `new_event` callbacks to start lagging by one.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7749>

2024-12-19 13:57:07 +0100  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* ext/gl/gstglmixerbin.c:
	  glmixerbin: derivate the mixer name from the bin name
	  Make it easier to identify a specific instance in logs in complex
	  pipelines.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8183>

2024-12-20 15:31:19 +0100  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* gst-libs/gst/gl/gstglcontext.c:
	  glcontext: add trace log when scheduling function in GL thread
	  There is already a log when the function is executed. Adding one when
	  scheduled can be useful to know how long we are waiting.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8190>

2025-01-04 12:20:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst/tcp/gsttcpclientsrc.c:
	* gst/tcp/gsttcpserversrc.c:
	  tcpclientsrc, tcpserversrc: don't try to get stats from NULL socket
	  Fixes criticals during gst-inspect.
	  Wasn't an issue before because the entire stats gathering
	  code path was inactive.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8211>

2024-12-24 18:24:08 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/tcp/gsttcpsrcstats.c:
	  tcpclientsrc, tcpserversrc: fix tcp stats gathering
	  Fix-up after commit e56b78417. Unclear how that could
	  ever have worked. Code only built because basically
	  everything was disabled due to the missing config.h
	  include.
	  - Include config.h so that HAVE_* defines are available.
	  - Fix missing variable declarations that somehow disappeared
	  when moving the stats gathering code block into a separate
	  function in a new file.
	  - Fix structure variable modified to match name in new function.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8211>

2025-01-03 14:16:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  message: Remove nullable annotation from gst_message_writable_details()
	  This function can never return NULL as the details are created for the message
	  if there are none yet.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8235>

2024-12-31 10:31:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/audio-channels.h:
	  audio: Add top-surround-left and right channel positions
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8171>

2024-12-18 19:45:27 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/audio-channels.c:
	* gst-libs/gst/audio/audio-channels.h:
	  audio: Add new gst_audio_reorder_channels_with_reorder_map()
	  This allows reordering the samples with a pre-calculated reorder map instead of
	  calculating it again every time.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8171>

2024-12-27 14:27:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Free main input even if it is not part of the list of inputs
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8216>

2024-12-23 09:49:51 +0100  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Gracefully handle missing data without prior input segment
	  Exit early in case the decoder has been resetted and hasn't received a new
	  segment event yet.
	  This was detected with netsim dropping buffers in a WebRTC pipeline.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8200>

2024-12-18 13:18:32 -0300  Thibault Saunier <tsaunier@igalia.com>

	* docs/meson.build:
	  doc: Handle gst_dep.get_variable('libexecdir') failure
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8178>

2024-12-18 12:27:30 -0300  Thibault Saunier <tsaunier@igalia.com>

	* docs/meson.build:
	  doc: Allow updating the plugins cache for all modules even if hotdoc is not present
	  This was possible for some modules but not all, for no good reason.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8178>

2023-04-18 11:34:54 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	* gst-libs/gst/pbutils/codec-utils.h:
	  pbutils: New AV1 caps utility functions
	  Makes it easy to convert between caps and codec_data
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4442>

2024-12-16 16:36:06 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Avoid deadlock on shutdown
	  The reason why the STATE lock was taken was to avoid issues where we would be
	  adding (and activating) elements at the same time as urisourcebin would be
	  brought down to READY. That would cause those new elements to potentially return
	  ERRORS because of not-negotiated/flushing-pads
	  But that creates a really bad deadlock (state lock is taken to deactivate the
	  streaming thread which .. is currently grabbing the state lock).
	  Instead, we can just ignore the warning/error messages that might occur when
	  shutting down.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4075
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8162>

2024-12-10 18:04:21 +0100  Robert Mader <robert.mader@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: Add DRM raw uploader
	  The goal is equivalent to adding DMA_DRM caps to the existing raw
	  uploader, but creating a new dedicated uploader allows separating the
	  logic more easily.
	  Test:
	  - use an AMD system
	  - modprobe visl
	  - gst-launch-1.0 filesrc location=some-1080p-content.mkv ! parsebin ! v4l2slh265dec ! glimagesink
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8126>

2024-12-13 15:51:20 +0300  Max Romanov <max.romanov@gmail.com>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: Decrease log level for item drop
	  Leaky appsrc feature was implemented similar to queue. Dropping item
	  is a normal case because it is configured so. Warning messages
	  is too severe for this event.
	  Level changed to DEBUG just like in queue.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8147>

2024-12-13 10:07:58 +0800  Dean Zhang (张安迪) <dean.zhang@mediatek.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: set decode only flag by decode only buffer
	  This is used for optimize frame drop issue. Some video streams such as av1
	  contains decode-only frames. These frames are handled by decoder but not show
	  by display, which will cause input frame count unmatch the output frame count.
	  After collecting more decoder only frame, it will cause error report "so many
	  frame droped". So attach the flag GST_BUFFER_FLAG_DECODE_ONLY with the input
	  buffer, the v4l2videodecoder can further handle the decode-only buffer in
	  output list.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8142>

2024-12-12 05:51:09 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Only rewrite stream-start event once
	  If the custom field is already present, we don't want to rewrite the event
	  again. Otherwise it would cause downstream into believing that a *new*
	  stream-start event has arrived ... which isn't the case.
	  Fixes #4097
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8132>

2024-12-12 10:43:26 +0100  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* gst-libs/gst/gl/gbm/gstgldisplay_gbm.c:
	  gl: raise WARNING instead of ERROR when no connector is connected
	  This is a valid and working use case when rendering on a headless
	  server.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8136>

2024-12-10 12:52:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  meson: unset GST_TRACERS for g-ir-scanner to avoid warnings
	  People might have GST_TRACERS=leaks set in their environment
	  by default, which will now trigger criticals during the build
	  when calling g-ir-scanner, because we unset GST_PLUGIN_SYSTEM_PATH
	  so that the scanner doesn't load any plugins.
	  Fixes #4093
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8121>

2024-11-28 11:57:57 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Don't use decodebin3 static sink pad
	  Makes the code leaner, and fixes the issue which would happen when the
	  urisourcebin stream that was linked to the sink pad goes away (without any
	  replacement).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7949>

2024-11-28 11:55:36 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Make usage of static sink pad optional
	  There is no reason why we should mandate people to "at least" use the static
	  sink pad. This caused issues, like mandating that it should always have valid
	  content linked to it (problematic in case of upstream stream changes).
	  Instead we only use it if it's actually linked to, in which case it gets added
	  to the list of inputs.
	  This actually simplifies the code too.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7949>

2024-11-25 10:50:34 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Fix issue re-using outputs
	  When checking whether a no-longer used output could be re-used for another slot,
	  we only want to do that for streams which are not still used.
	  Otherwise we end up potentially re-assigning a demuxer stream to a completely
	  different one
	  Furthermore, if we *are* re-using an output slot, indicate what the replacement
	  GstStream will be so slot matching can work properly (which can happen in the
	  case of demuxers which add/remove all pads even if only a single one changed)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7949>

2024-12-03 15:16:53 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/meson.build:
	  build: gl: Properly add sub-package to gst_libraries
	  GstGLEGL/Wayland/X11/Viv was actually never written into any of the gir_targets,
	  as a side effect, their gir was never copied into girs/ and not used to generate
	  the documentation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8061>

2024-12-04 18:45:30 +0800  Cheung Yik Pang <pang.cheung@harmonicinc.com>

	* gst-libs/gst/video/video-format.c:
	  video: fix AV12 format lacking the GST_VIDEO_FORMAT_FLAG_ALPHA flag
	  The GST_VIDEO_FORMAT_FLAG_ALPHA indicates whether a video format has an
	  alpha channel.  AV12 does have an alpha plane.  Switching from
	  MAKE_YUV_FORMAT to MAKE_YUVA_FORMAT to generate GST_VIDEO_FORMAT_AV12.
	  Fixes #4081
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8067>

2024-12-05 09:21:04 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Fix collection leak
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8080>

2024-11-20 14:16:23 +0100  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* gst-libs/gst/video/video-sei.c:
	  h264parse: parse unregistered SEI without user data
	  We get loads of warnings when parsing videos from users:
	  gsth264parser.c:1115:gst_h264_parser_parse_user_data_unregistered: No more remaining payload data to store
	  gsth264parse.c:646:gst_h264_parse_process_sei:<h264parse0> failed to parse one or more SEI message
	  Those are raised because of unregistered SEI without user data.
	  The spec does not explicitly state that unregistered SEI needs to have
	  data and I suppose the UUID by itself can carry valuable information.
	  FFmpeg also parses and exposes such SEI so there is no reason for us no
	  too as well.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7931>

2024-09-30 21:35:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/vorbis/gstvorbisdec.c:
	  vorbisdec: Set at most 64 channels to NONE position
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-115
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3869
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8035>

2024-10-09 11:23:47 -0400  Sebastian Dröge <sebastian@centricular.com>

	* gst/subparse/gstsubparse.c:
	  subparse: Check for NULL return of strchr() when parsing LRC subtitles
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-263
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3892
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8039>

2024-09-30 18:36:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/subparse/gstssaparse.c:
	* meson.build:
	  ssaparse: Don't use strstr() on strings that are potentially not NULL-terminated
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8036>

2024-09-30 21:40:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/subparse/gstssaparse.c:
	  ssaparse: Search for closing brace after opening brace
	  Otherwise removing anything between the braces leads to out of bound writes if
	  there is a closing brace before the first opening brace.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-228
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3870
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8036>

2024-09-30 18:19:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tools/gst-discoverer.c:
	  discoverer: Don't print channel layout for more than 64 channels
	  64+ channels are always unpositioned / unknown layout.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-248
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3864
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8034>

2024-10-02 16:52:51 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/ogg/gstoggstream.c:
	  oggstream: review and fix per-format min_packet_size
	  This addresses all manually detected invalid reads in setup functions.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8038>

2024-10-02 15:16:30 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* ext/ogg/vorbis_parse.c:
	  vorbis_parse: check writes to GstOggStream.vorbis_mode_sizes
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-117 Fixes gstreamer#3875
	  Also perform out-of-bounds check for accesses to op->packet
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8038>

2024-10-01 13:22:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/opus/gstopusdec.c:
	  opusdec: Set at most 64 channels to NONE position
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-116
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3871
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8037>

2024-09-26 13:43:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/tag/id3v2.c:
	  id3v2: Don't try parsing extended header if not enough data is available
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-235
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3842
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8033>

2024-11-29 20:57:09 +0100  Jakub Adam <jakub.adam@collabora.com>

	* ext/gl/gstgldownloadelement.c:
	  gldownload: warn on fallback from DMABuf to system memory
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8013>

2024-11-26 15:42:12 -0500  Xavier Claessens <xclaessens@netflix.com>

	* gst/videorate/gstvideorate.c:
	* tests/check/elements/videorate.c:
	  videorate: convert next_ts to new segment instead of restarting from 0
	  When receiving a new segment we should not restart PTS from the new
	  segment' start. Instead convert current position into the new segment if
	  possible.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7977>

2024-11-26 13:52:03 -0500  Xavier Claessens <xclaessens@netflix.com>

	* gst/audiorate/gstaudiorate.c:
	* gst/audiorate/gstaudiorate.h:
	* tests/check/elements/audiorate.c:
	  audiorate: convert next_ts to new segment instead of restarting from 0
	  When receiving a new segment we should not restart PTS from the new
	  segment' start. Instead convert current position into the new segment if
	  possible.
	  Fixes: #4060
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7977>

2024-11-27 09:26:15 -0500  Xavier Claessens <xclaessens@netflix.com>

	* gst/audiorate/gstaudiorate.c:
	  audiorate: Always push updated segment
	  Convert segment to TIME format immediately instead of waiting for
	  _chain() to be called. This fixes converted segment never being pushed
	  downstream.
	  Fix the convert function that was copying some fields in the wrong
	  direction. Add fast copy if segment is already in TIME format.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7977>

2024-11-29 17:33:29 +0000  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.c:
	* tests/check/libs/rtp.c:
	  rtp: Fix precision loss in gst_rtcp_ntp_to_unix()
	  Without this patch the UNIX timestamp resulting from the translation from NTP
	  would be off by one nano-second.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8010>

2024-12-01 11:39:57 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtsp: Fix off-by-one in month/day-of-the-week array
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8017>

2024-11-29 09:42:41 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/gl/egl/gsteglimage.c:
	  gl: Add missing Since marker
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8001>

2024-07-29 10:41:42 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* ext/gl/gstgldmabufbufferpool.c:
	  gstgldmabufbufferpool: use gsteglimagecache
	  Store the imported GstEGLImage in a GstEGLImageCache. Since this passes
	  ownership to the cache, stop unreffing the images from the GstMemory.
	  Free the cache when the buffer pool is stopped.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-07-29 10:24:44 +0200  Philipp Zabel <p.zabel@pengutronix.de>

	* gst-libs/gst/gl/egl/gsteglimagecache.c:
	* gst-libs/gst/gl/egl/gsteglimagecache.h:
	* gst-libs/gst/gl/gstglupload.c:
	* gst-libs/gst/gl/meson.build:
	  glupload/egl: Extract EGL image cache and make it public
	  Extract the EGLImage cache from gstglupload.c, place it in a separate
	  source file gsteglimagecache.c, and make the API public, so it can be
	  reused by the gldownload element.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-11-07 15:37:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gl/gstgldmabufbufferpool.c:
	* ext/gl/gstgldmabufbufferpool.h:
	* ext/gl/gstgldownloadelement.c:
	* ext/gl/gstgldownloadelement.h:
	  gldownload: Implement importing RGBA with modifiers
	  For single plane RGBA based formats, implement support for modifiers.
	  This fixes issues where linear is actually configured in GL causing
	  visual artifacts.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-09-11 18:31:10 +0200  Jakub Adam <jakub.adam@collabora.com>

	* ext/gl/gstgldownloadelement.c:
	* gst-libs/gst/gl/egl/gsteglimage.c:
	* gst-libs/gst/gl/egl/gsteglimage.h:
	* gst-libs/gst/gl/gstglutils.c:
	* gst-libs/gst/gl/gstglutils.h:
	  gldownload: enumerate emulated video formats
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-08-31 00:17:23 +0200  Jakub Adam <jakub.adam@collabora.com>

	* gst-libs/gst/gl/egl/gstglcontext_egl.c:
	  glcontext/egl: Fix use of uninitialized memory
	  At least some Mesa drivers (see llvmpipe_query_dmabuf_modifiers()) don't
	  initialize the provided external_only array, so some format/modifier
	  combinations could get incorrectly marked as only external.
	  Make sure we zerofill the array before passing it to
	  eglQueryDmaBufModifiersEXT().
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-08-29 14:37:36 +0200  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/gl/gstgldmabufbufferpool.c:
	  gldmabufbufferpool: allow the dmabuf_pool configuration to change
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-09-07 01:11:35 +0200  Jakub Adam <jakub.adam@collabora.com>

	* ext/gl/gstgldownloadelement.c:
	  downloadelement: support dma_drm caps in try_export_dmabuf
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-09-27 14:17:34 +0200  Jakub Adam <jakub.adam@collabora.com>

	* ext/gl/gstgldmabufbufferpool.c:
	* ext/gl/gstgldmabufbufferpool.h:
	* ext/gl/gstgldownloadelement.c:
	* ext/gl/gstgldownloadelement.h:
	* ext/gl/meson.build:
	  gldownload: import DMABufs from downstream pool
	  * if downstream provides us with a pool that can allocate DMA buffers,
	  gldownload wraps it into a GL buffer pool that uses
	  glEGLImageTargetTexture2DOES() to import dmabuf into textures
	  * upstream GL elements can allocate from that pool
	  * gldownload unwraps DMA buffers from incoming GL buffers and passes
	  them downstream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-09-02 20:46:23 +0200  Jakub Adam <jakub.adam@collabora.com>

	* gst-libs/gst/gl/egl/gstglcontext_egl.c:
	* gst-libs/gst/gl/egl/gstglcontext_egl.h:
	* gst-libs/gst/gl/gstglupload.c:
	  glcontext_egl: add method to check if format supports a modifier
	  Makes _check_modifier from glupload reusable.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-08-29 17:22:53 +0200  Jakub Adam <jakub.adam@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	* gst-libs/gst/gl/gstglutils.c:
	* gst-libs/gst/gl/gstglutils.h:
	  glupload: make _dma_buf_transform_* functions public API
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-08-29 17:10:54 +0200  Jakub Adam <jakub.adam@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: rename GstGLUploadDrmFormatFlags -> GstGLDrmFormatFlags
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6792>

2024-11-25 12:27:03 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Only send GAP events out of source pads
	  If streamsynchronizer is waiting on the stream's sinkpad and srcpad at the same
	  time, it can happen that the GAP event is otherwise sent out of the sinkpad,
	  which is in the wrong direction.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7950>

2024-11-27 19:11:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: fix build with older glib versions
	  Follow-up to !7969
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7988>

2024-11-26 15:35:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Fix formatting of a debug log string
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7969>

2024-11-26 15:32:56 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/pbutils/pbutils-private.h:
	  discoverer: Move cache file tracking from GstDiscovererInfo into GstDiscoverer
	  It's only used temporarily during discovery and makes no sense as part of the
	  resulting info.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7969>

2024-11-26 15:23:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/pbutils/pbutils-private.h:
	  discoverer: Move stream counter from GstDiscovererInfo into GstDiscoverer
	  It's only used temporarily during discovery and makes no sense as part of the
	  resulting info.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7969>

2024-11-26 15:20:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	  discoverer: Also copy missing element details when copying a GstDiscovererInfo
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7969>

2024-11-26 15:41:24 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstdiscoverer-types.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Make sure the missing elements details array is NULL-terminated in a thread-safe way
	  And also avoid a possible out-of-bounds read if the array contains no elements.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7969>

2024-11-26 06:40:34 +0100  Tomáš Polomský <1155369-polomsky@users.noreply.gitlab.freedesktop.org>

	* gst-libs/gst/app/gstappsink.c:
	  appsink: fix timeout logic for gst_app_sink_try_pull_sample
	  In case of gst_app_sink_try_pull_object returns an object instead of a sample,
	  the whole process must be restarted with the reduced timeout, otherwise requested
	  timeout could be easily exceeded.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7962>

2024-11-22 11:50:05 +0100  Edward Hervey <bilboed@bilboed.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Unify collection switching checks
	  We only want to switch to a selection of an output collection if all streams are
	  present.
	  This was previously only done in one place (when triggering by new incoming
	  streams) but not when triggered by user/application.
	  Avoid this by moving the check to handle_stream_switch()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7941>

2024-11-20 13:32:20 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* tests/check/libs/discoverer.c:
	  discoverer: fix segfault in race condition adding a new uri's
	  There is a race condition adding new uri's right after receiving
	  the `discovered` event. We must wait until we have cleaned-up
	  the last discovery to start processing the new one
	  Fix #3758
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7930>

2024-11-20 04:23:32 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/gl/meson.build:
	  meson: Don't use libdrm_dep in cc.has_header()
	  It can't handle subproject dependencies. Pointed out in
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4033#note_2665974
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7927>

2024-11-14 15:56:50 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: dmabuf: Only pass padded size for indirect tiled handling
	  We only require out_info to hold on padded width/height when
	  using our internal detiling shaders. Doing so for direct upload
	  leads to the image being scaled down in a way that the padding
	  is displayed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7849>

2024-11-07 17:25:20 +0100  Benjamin Gaignard <benjamin.gaignard@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	  video: dma-drm: Add DRM vendor modifiers list
	  Copy DRM vendor modifier IDs from drm_fourcc.h
	  Add Vivante modifier definition and map GST_VIDEO_FORMAT_NV12_4L4 and
	  GST_VIDEO_FORMAT_NV12_10LE40_4L4 accordingly. Also add Samsung 64Z32 tiled
	  format mapping.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7849>

2024-11-06 18:08:30 +0100  Benjamin Gaignard <benjamin.gaignard@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	  video: dma-drm: Add NV15, NV20 and NV30 DRM formats definitions
	  Cherry pick DRM NV15, NV20 and NV30 formats definitions from drm_fourcc.h
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7849>

2024-11-13 15:05:12 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	  video: dma drm: Fix modifier definition on 32bit
	  The modifiers are 64bit, add missing ULL suffix so that the compiler does
	  not complain on 32bit platforms.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7849>

2024-11-07 17:08:16 +0100  Benjamin Gaignard <benjamin.gaignard@collabora.com>

	* gst-libs/gst/video/video-info-dma.c:
	* gst-libs/gst/video/video-info-dma.h:
	  video: dma-drm: Add modifier field to FormatMap
	  A GST_VIDEO_FORMAT_* could be mapped to only one pair of DRM_FORMAT_*
	  and DRM_FORMAT_MOD_*. Until now only DRM_FORMAT_MOD_LINEAR was used.
	  To be able to add other modifiers add a modidier field in struct FormatMap.
	  Create a helper functions the allow turning a GstVideoFormat into a pair
	  of DRM fourcc and modifier and vis-versa.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7849>

2024-10-10 15:51:36 -0400  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/sdp/gstmikey.c:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	  sdp: Add debug categories for message and mikey modules
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7924>

2024-09-19 20:57:09 +0800  Qian Hu (胡骞) <qian.hu@mediatek.com>

	* gst-libs/gst/video/gstvideometa.h:
	  docs: update fields about gst video aligment
	  1) add stride align fields in allocation.md
	  2) add stride align fields in gstvideometa.h
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7201>

2024-08-19 13:50:48 +0800  Qian Hu (胡骞) <qian.hu@mediatek.com>

	* gst-libs/gst/video/gstvideometa.c:
	  meta: add aggregation function for allocation meta api params
	  since tee do allocation query for each downstream element, and
	  get allocation query params from them. this function will aggregate
	  all params, and propose it for upstream element of tee
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7201>

2024-11-18 18:48:47 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/video/gstvideopool.c:
	* gst-libs/gst/video/gstvideopool.h:
	  video: pool: Make structure const when getting video alignment
	  Downgrade to const, this is backward compatible make makes the API compatible
	  wit more usages.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7201>

2024-11-12 09:44:30 -0300  Thibault Saunier <tsaunier@igalia.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst-libs/gst/gl/gstglcolorconvert.c:
	* gst-libs/gst/gl/gstglcolorconvert.h:
	* gst-libs/gst/gl/gstglformat.c:
	* gst-libs/gst/gl/gstglmemory.h:
	  gl: colorconvert: Add support for I422_10XX and I422_12XX
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7919>

2024-11-13 16:59:18 +1100  Matthew Waters <matthew@centricular.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebin: silence a maybe-unitialized warning
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7875>

2024-11-13 16:57:42 +1100  Matthew Waters <matthew@centricular.com>

	* ext/ogg/gstoggdemux.c:
	  oggdemux: silence a maybe-unitialized warning
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7875>

2024-11-16 18:24:23 +0000  Philippe Normand <philn@igalia.com>

	* gst/playback/gstplaybackutils.c:
	  playbackutils: Fix caps leak in get_n_common_capsfeatures()
	  The gst_static_caps_get() return value is transfer-full.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7908>

2024-11-14 15:15:43 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: Upgrade uploader selection message to INFO
	  This allow using INFO log to quickly see what uploader we
	  picked. This is very useful as the DEBUG category has became
	  very noisy.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7891>

2024-11-14 15:13:50 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: dmabuf: Fix emulated tiled import
	  The added check for LINEAR modifier broke dmabuf import for
	  NV12 emulation (R8/RG88) of all sort (linear and tiled).
	  Fixes !5461
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7891>

2024-11-15 11:46:14 -0300  Thibault Saunier <tsaunier@igalia.com>

	* meson.build:
	  meson: Bump minimum version to 1.3
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4025
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7899>

2024-11-14 15:58:58 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiostreamalign.c:
	  audiostreamalign: Don't report disconts for every buffer if alignment-threshold is too small
	  If it is too small, the maximum allowed diff becomes 0 samples which would then
	  trigger if there is no discontinuity at all.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7888>

2024-11-07 23:48:15 -0300  Thibault Saunier <tsaunier@igalia.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst-libs/gst/gl/gstglcolorconvert.c:
	* gst-libs/gst/gl/gstglcolorconvert.h:
	* gst-libs/gst/gl/gstglformat.c:
	* gst-libs/gst/gl/gstglmemory.h:
	  gl: colorconvert: Add support for Y444_10XX and Y444_16XX
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7858>

2024-11-04 10:37:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: Add support for M101 Matrox uncompressed SD video codec
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7823>

2023-10-28 13:01:58 +0200  Diego Nieto <diego.nieto.m@outlook.com>

	* gst-libs/gst/tag/gstexiftag.c:
	* gst-libs/gst/tag/gsttageditingprivate.c:
	* gst-libs/gst/tag/gsttageditingprivate.h:
	* gst-libs/gst/tag/tag.h:
	* gst-libs/gst/tag/tags.c:
	  exiftag: handle GST_TAG_CAPTURING_LIGHT_SOURCE tag
	  This exif tag allows to specify the different light conditions
	  when taking a picture. This tag is defined in:
	  https://exiftool.org/TagNames/EXIF.html#LightSource
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5571>

2024-11-04 16:54:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/audioresample.c:
	  audioresample: skip pointless loop in broken test_fft unit test
	  Variable f1 is never used, so just skip that loop for now.
	  The test has never actually tested actual resampling because of
	  that bug it seems, and the test fails if fixed to actually resample.
	  For now we just avoid the pointless 126*12 pipelines that were just
	  testing the same thing (nothing) over and over again.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7827>

2024-11-04 09:21:44 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Also use event probe for HLS use-cases
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3957
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7822>

2024-11-06 18:51:21 +1100  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/video/convertframe.c:
	  video-converter: Set TIME segment format on appsrc
	  Combine the appsrc and appsink settings into one place and ensure that
	  the appsrc will output a TIME segment, to avoid incorrect segment format
	  criticals in some situations.
	  The D3D11 path was already setting the segment format correctly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7839>

2024-11-05 00:01:56 +0100  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst-libs/gst/gl/cocoa/gstglwindow_cocoa.m:
	  cocoa: fix rendering artefacts in retina displays
	  Set the correct contentsScale in the CALayer instead of using
	  the default (1.0) which causes rendering artefacts in retina
	  displays due to using a smaller bitamp than the display size.
	  See: https://developer.apple.com/documentation/quartzcore/calayer/1410746-contentsscale?language=objc
	  Fix: #3942
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7828>

2024-11-03 17:30:40 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/meson.build:
	  meson: bail out earlier in docs subdir if docs are disabled
	  The gst_dep.get_variable('libexecdir') may fail in some scenarios
	  (e.g. building a module alone inside an uninstalled devenv) and
	  it shouldn't really be reached in the first place if docs are
	  disabled via options.
	  Also to avoid confusing meson messages when cross-compiling or
	  doing a static build.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7818>

2024-11-03 17:36:38 +0000  Tim-Philipp Müller <tim@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/fur.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/ka.po:
	* po/lt.po:
	* po/lv.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  gst-plugins-base: update translations
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7819>

2024-06-13 09:59:24 +0200  Andreas Wittmann <andreas.wittmann@zeiss.com>

	* gst-libs/gst/gl/glprototypes/all_functions.h:
	* gst-libs/gst/gl/glprototypes/invalidate_subdata.h:
	* gst-libs/gst/gl/gstglfuncs.h:
	* gst-libs/gst/gl/meson.build:
	  qmlgloverlay: make depth buffer optional
	  allow dropping depth/stencil buffer for
	  qmlgloverlay element. Shows considerable
	  drop in memory bandwidth consumption of
	  the element on windows platforms.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7677>

2024-10-25 10:38:36 +0200  Albert Sjolund <alberts@axis.com>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: Fix use-after-free when making buffer / buffer-lists writable
	  make_writable can cause a reallocation of the buffer, meaning that obj
	  would point to an invalid object, both for buffer and for bufferlist.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7803>

2024-10-25 21:46:45 +1100  Jan Schmidt <jan@centricular.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: Check for memory capsfeature on video/x-raw
	  Don't assume that video/x-raw caps means buffers are mappable
	  or can be processed by videoconvert and friends. Only plug
	  those converters for real system memory, and treat other
	  memory capsfeatures as hardware surfaces
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7741>

2024-10-28 12:38:39 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Fix typos
	  discovererd / discoverered -> discovered
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7733>

2024-10-22 19:04:40 -0300  Thibault Saunier <tsaunier@igalia.com>

	* gst-libs/gst/pbutils/gstdiscoverer.c:
	  discoverer: Fix serializing tags StreamInfo on disc
	  When `use-cache` is used, we were serializing the StreamInfo **before**
	  _ensure_info_tags was called, meaning we were always missing that information.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7733>

2024-10-23 15:39:17 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	* gst/playback/gstplaybackutils.c:
	* gst/playback/gstplaybackutils.h:
	* gst/playback/gsturisourcebin.c:
	  playback: Improve stream list search
	  There is the possibility than an element/code/helper creates an identical
	  `GstStream` (same type and stream-id) instance instead of re-using a previous
	  one.
	  For those cases, when detecting whether a `GstStream` is already present in a
	  collection, we need to do more checks than just comparing the pointer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7716>

2024-10-23 15:17:22 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Aggregate collections from multiple parsebin
	  In the case where multiple parsebin are present (ex: from rtsp sources), we want
	  to aggregate the collections provided by the different parsebin and expose a
	  single "unified" collection.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7716>

2024-10-23 14:25:52 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Store caps on parsepad as early as possible
	  When analyzing a new pad, we can store the updated caps on the GstStream if they
	  are fixed.
	  If they are not fixed, this function will be called again once the element will
	  provide them.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7716>

2024-10-23 14:24:36 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Don't store bogus caps on GstStream
	  If we can't get the current caps when receiving a stream-start, that's fine,
	  they can/will be provided by other means at a later time.
	  What we definitely should not do is provide the starting caps of the chain,
	  which are potentially completely different from the end ones (like for example
	  `application/x-rtp`)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7716>

2024-10-23 14:24:18 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Minor debug statement updates
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7716>

2024-10-15 15:58:53 +0100  Philippe Normand <philn@igalia.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebasebin: Fix parser selection
	  If an encoder supports multiple codecs (a bin wrapping/auto-plugging encoders)
	  then its src pad template caps might list the supported codecs. Without this
	  patch the selected parser would be the one corresponding to the first codec,
	  leading to caps negotiation error later on. The proposed fix is to check the
	  media type on the parser candidates sink pad templates according to the
	  requested encoded format.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7670>

2024-10-15 15:57:06 +0100  Philippe Normand <philn@igalia.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebasebin: Fix parser cleanup
	  When removing the parser from the stream group, it might not be linked to a
	  combiner so this needs to be checked.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7670>

2024-10-19 12:15:00 +0200  Diego Nieto <diego.nieto.m@outlook.com>

	* gst-libs/gst/video/video-frame.c:
	  video-format: reduce the number of memcpy if possible
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7694>

2024-10-24 09:17:54 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/gl/gstglcontext.c:
	* gst-libs/gst/gl/gstgldisplay.c:
	* gst-libs/gst/gl/gstglwindow.c:
	* gst-libs/gst/pbutils/install-plugins.c:
	  all: Fix closure annotations
	  This was misused almost everywhere.
	  See
	  https://gi.readthedocs.io/en/latest/annotations/giannotations.html#support-for-gobject-closures
	  and: https://www.bassi.io/articles/2023/02/20/bindable-api-2023/
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7725>

2024-10-21 11:50:23 +0200  Carlos Falgueras García <cfalgueras@fluendo.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst-libs/gst/video/video-converter.c:
	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-format.h:
	* gst-libs/gst/video/video-info.c:
	  video: Add GRAY10_LE16 support
	  This adds a 10-bit variant of grayscale packed into 16 bits little-endian
	  words. The MSB 6 bits are padding and should be ignored. This format is
	  used by Fraunhofer VVC encoder and decoder libraries.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7706>

2024-10-23 20:50:31 +0200  François Laignel <francois@centricular.com>

	  rtspsrc: client-managed MIKEY KeyMgmt
	  Some servers (e.g. Axis cameras) expect the client to propose the encryption
	  key(s) to be used for SRTP / SRTCP. This is required to allow re-keying so
	  as to evade cryptanalysis. Note that the behaviour is not specified by the
	  RFCs. By setting the 'client-managed-mikey-mode' property to 'true', rtspsrc
	  acts as follows:
	  * For a secured profile (RTP/SAVP or RTP/SAVPF), any media in the SDP
	  returned by the server for which a MIKEY key management applies is
	  elligible for client managed mode. The MIKEY from the server is then
	  ignored.
	  * rtspsrc sends a SETUP with a MIKEY payload proposed by the user. The
	  payload is formed by calling the 'request-rtp-key' signal for each
	  elligible stream. During initialisation, 'request-rtcp-key' is also
	  called as usual. The keys returned by both signals should be the same
	  for a single stream, but the mechanism allows a different approach.
	  * The user can start re-keying of a stream by calling SET_PARAMETER.
	  The convenience signal 'set-mikey-parameter' can be used to build a
	  'KeyMgmt' parameter with a MIKEY payload.
	  * After the server accepts the new parameter, the user can call
	  'remove-key' and prepare for the new key(s) to be served by signals
	  'request-rtp-key' & 'request-rtcp-key'.
	  * The signals 'soft-limit' & 'hard-limit' are called when a key
	  reaches the limits of its utilisation.
	  This commit adds support for:
	  * client-managed MIKEY mode to srtpsrc.
	  * Master Key Index (MKI) parsing and encoding to GstMIKEYMessage.
	  * re-keying using the signals 'set-mikey-parameter' & 'remove-key' and
	  then by serving the new key via 'request-rtp-key' & 'request-rtcp-key'.
	  * 'soft-limit' & 'hard-limit' signals, similar to those provided by srtpdec.
	  See also:
	  * https://www.rfc-editor.org/rfc/rfc3830
	  * https://www.rfc-editor.org/rfc/rfc4567
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7587>

2024-10-23 11:49:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Remove "source" property
	  There is not necessarily a single urisourcebin inside uridecodebin3 so having a
	  single source property makes little sense. Additionally, this property was never
	  hooked up at all and always returned NULL.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7712>

2024-10-22 13:23:05 +0100  James Cowgill <james.cowgill@blaize.com>

	* gst-libs/gst/tag/gstexiftag.c:
	  exiftag: Check the result of gst_date_time_new_local_time
	  Fixes a critical warning when parsing a JPEG containing a malformed
	  `DateTime` tag (like `0000:00:00 00:00:00`).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7709>

2024-10-18 23:51:53 -0600  Scott Moreau <oreaus@gmail.com>

	* gst-libs/gst/gl/egl/gstglcontext_egl.c:
	  gst-plugins-base: Set swap interval to 0 for wayland EGL windows
	  This allows the stream to drive the buffers submitted to the display server.
	  If the application does not receive frame events for a period of time due to
	  minimization or tty switch for example, instead of waiting to process and
	  then catching up when frame events resume, the stream will resume instantly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7691>

2024-10-18 14:41:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/egl/gstglcontext_egl.c:
	  glcontext: egl: Unrestrict the support base DRM formats
	  There is no requirement for a base DRM format to be supported by libgstvideo
	  in order to be uploaded to. Don't limite to DRM fourcc that have a libgstvideo
	  format mapping. This notably enabled AFBC support, which uses an opaque based
	  format that does not have a linear definition. This also adds R8/RG88 and
	  simimlar other formats that are not yet mapped in libgstvideo.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7689>

2024-10-17 01:03:04 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/playback/gstplaybin2.c:
	* gst/playback/gstplaybin3.c:
	  playback: remove mention of gconf sinks from playbin docs
	  Fixes #3916
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7676>

2024-10-16 17:20:46 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Don't remove probe if not present
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>

2024-10-03 17:16:25 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Set pad probe before linking pad
	  We want to grab all events that pass through, so we need to set the
	  probe **before** the pad is linked
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>

2024-10-03 17:16:08 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Streamline event forwarding code
	  Instead of two different functions for copying events and rewriting the
	  stream-start event, just have a single one.
	  This is needed, since we want to do both of those in one go, with the pad lock
	  taken.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>

2024-10-03 17:09:13 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Set stream collection on pad before exposing it
	  We want to ensure the stream-collection is present on the pad (as a sticky
	  event) before we expose the pad.
	  This is more reliable since it will ensure it is present before any other event
	  is pushed through.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>

2024-10-03 17:07:59 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Use lock when checking input/output items
	  Otherwise there is the risk that the play items list could have changed in
	  between
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>

2024-10-03 17:05:15 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Free pending events on input pads when unlinking
	  Otherwise we could end up with *old* events leaking through on the next linking.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7609>

2024-10-09 15:45:34 -0400  François Laignel <francois@centricular.com>

	* gst-libs/gst/audio/audio-info.c:
	* gst-libs/gst/audio/gstaudioaggregator.c:
	* gst-libs/gst/video/video-info.c:
	  gst: hamonise recent Caps function names with Structure counterparts
	  These functions were introduced as part of the GstIdStr MR:
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7644>

2024-10-14 15:51:49 +0200  Diego Nieto <diego.nieto.m@outlook.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: descriptions: add H.266/VVC descriptions
	  Add H.266 descriptions supporting different codec profiles as well
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7652>

2024-10-11 18:06:38 +0200  Diego Nieto <diego.nieto.m@outlook.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: add H.266/vvc mime type
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7652>

2024-08-10 18:53:41 +0200  Ruben Gonzalez <rgonzalez@fluendo.com>

	  typefind: Add typefinder for VVC/H.266
	  H.266 NAL unit header syntax [1] is similar to H.265 NAL unit header syntax[2]:
	  ```
	  H.265                               H.266
	  +---------------+---------------+   +---------------+---------------+
	  |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|   |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
	  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
	  |F| NALType   | LayerId   | TID |   |F|U| LayerId | NALType   | TID |
	  +-------------+-----------+-----+   +-------------+-----------------+
	  Where
	  * F: `forbidden_zero_bit`: f(1)
	  * U: `nuh_reserved_zero_bit`: u(1) only H.266
	  * LayerId: `nuh_layer_id`: u(6)
	  * NALType: `nal_unit_type`: u(6) in H.265 and u(5) in H.266
	  * TID: `nuh_temporal_id_plus1`: u(3)
	  ```
	  NAL unit types have different values:
	  | NALType  | H.265                              | H.266                     |
	  |----------|------------------------------------|---------------------------|
	  | VPS      | HEVC_NAL_VPS(32)                   | VVC_VPS_NUT(14)           |
	  | SPS      | HEVC_NAL_SPS(33)                   | VVC_SPS_NUT(15)           |
	  | PPS      | HEVC_NAL_PPS(34)                   | VVC_PPS_NUT(16)           |
	  | IRAP     | BLA_W_LP(19)..HEVC_NAL_CRA_NUT(21) | IDR_W_RADL(7)..CRA_NUT(9) |
	  Implementation of `h266_video_type_find` is based on `h265_video_type_find` with
	  next differences:
	  - NAL unit header syntax for H.265 and H.266
	  - Diff NAL unit types values
	  - Avoid checking nuh_layer_id is zero.  H.266 conformance test suite[3] contains examples with more than one layer.
	  This typefind was tested with H.266 conformance test suite [3]. Also, with the help of fluster[4],
	  with H.264 and H.265 conformance test suites to avoid regresions. Pending test vectors to fix:
	  - 8b422_H_Sony_4
	  - DEBLOCKING_E_Ericsson_3
	  [1] https://www.itu.int/rec/T-REC-H.266
	  [2] https://www.itu.int/rec/T-REC-H.265
	  [3] https://www.itu.int/wftp3/av-arch/jvet-site/bitstream_exchange/VVC/draft_conformance/draft6/
	  [4] https://github.com/fluendo/fluster/
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7339>

2024-08-11 15:25:58 +0200  Ruben Gonzalez <rgonzalez@fluendo.com>

	* gst/typefind/gsttypefindfunctions.c:
	  misc: fix grammar mistake
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7339>

2024-05-12 18:42:21 -0400  valadaptive <valadaptive@protonmail.com>

	* gst-libs/gst/riff/riff-media.c:
	  riff: Add support for Lagarith fourcc tag
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6831>

2024-09-27 15:09:38 +0200  Théo Maillart <tmaillart@freebox.fr>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: do not attempt to remove a null stream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7586>

2024-09-26 11:29:08 +0200  Théo Maillart <tmaillart@freebox.fr>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: protect internal reset with SELECTION_LOCK
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7586>

2024-09-25 15:32:20 +0200  Théo Maillart <tmaillart@freebox.fr>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: remove output event probe on remove input stream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7586>

2024-10-02 11:29:38 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Ensure all stream-start are handled
	  In order to ensure all initial events (stream-start, caps, ..) are present on
	  pads that we expose, those various sticky events are propagated (from parsebin
	  to multiqueue output, from multiqueue output to exposed pads).
	  The problem was that the "hack" in `urisourcebin` to inform downstream elements
	  that the stream is parsed data and a collection will be present was only done in
	  one place : a probe on the output of parsebin ... but the stream-start could
	  potentially have already been propagated to the output pads before that.
	  In order to fix that, we make sure any pending sticky stream-start event is
	  updated before being propagated.
	  Fixes #3788
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7598>

2024-09-04 15:10:30 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: Add LCEVC information to H.264 description
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>

2024-10-02 10:34:57 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Allow more cases for posting stream-collection
	  Previously urisourcebin only allows stream-collections messages from adaptive
	  demuxers or sources to be posted.
	  This commit also allows the case where they come from a single parsebin. We
	  still want to prevent it in the case where they are multiple parsebins, since
	  that would require some form of aggregation to show a single/unified collection.
	  In order to avoid a regression with uridecodebin3 behavior, we also implement
	  support for GST_QUERY_SELECTABLE, so that uridecodebin3 can figure out whether
	  it should let GST_MESSAGE_STREAM_COLLECTION flow upwards (because app/user could
	  react on it) or whether it drops it in order for decodebin3 to do the collection
	  aggregation and posting.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7597>

2024-10-01 17:17:01 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Make update/posting of collection messages atomic
	  The presence (or not) of a collection on an input will determine whether events
	  will be throttled so that there are only forwarded when that input gets a valid
	  collection.
	  Therefore the input lock should be used.
	  In addition to that, we want to ensure that the application/user has a chance to
	  reliably (i.e. synchronously) specify what streams it is interested in by
	  sending a GST_EVENT_SELECT_STREAMS.
	  But we cannot allow anything to go forward until that message posting has come
	  back, otherwise we run in various races.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3872
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7594>

2024-09-27 19:45:06 +0900  Elliot Chen <elliot.chen@nxp.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: check and send selected stream message even if no decoder is selected
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7579>

2024-09-27 09:23:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/allocators/gstdrmdumb.c:
	  allocators: drmdump: Add NV15 supports
	  This is a packed version of P010, names NV14_10LE40 in GStreamer. This
	  format is used on all Rockchip SoC variants.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7580>

2024-09-27 09:20:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/allocators/gstdrmdumb.c:
	  allocators: drmdumb: Fix bpp value for P010
	  P010 uses 16 bits per pixel, with least significant being padding. This
	  code worked with Intel display driver since they roundup that value, but
	  does not work with the generic DRM helpers which also support NV15,
	  which does not have any padding.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7580>

2024-09-03 13:19:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-format.c:
	* gst-libs/gst/audio/audio-info.c:
	* gst-libs/gst/audio/gstaudioaggregator.c:
	* gst-libs/gst/video/gstvideoaggregator.c:
	* gst-libs/gst/video/video-converter.c:
	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-info.c:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audiomixer/gstaudiointerleave.c:
	* gst/audioresample/gstaudioresample.c:
	* gst/compositor/compositor.c:
	* gst/videoconvertscale/gstvideoconvertscale.c:
	  base: audio: video: Use more efficient caps/structure API in various places
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432>

2024-09-03 12:16:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/gl/gstgldownloadelement.c:
	* ext/gl/gstglimagesink.c:
	* ext/gl/gstglstereomix.c:
	* ext/gl/gstglviewconvert.c:
	* gst-libs/gst/gl/gstglupload.c:
	* gst-libs/gst/video/video-info-dma.c:
	* gst-libs/gst/video/video-info.c:
	* gst/videoconvertscale/gstvideoconvertscale.c:
	* tests/check/libs/gstglcolorconvert.c:
	* tests/check/libs/gstglupload.c:
	  common: Use more efficient versions of GstCapsFeatures API where possible
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432>

2024-08-30 18:57:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/gl/gstglupload.c:
	  common: Stop using GQuark-based GstCapsFeatures API
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432>

2024-08-29 20:09:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/gl/gstglfiltershader.c:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/rtp/gstrtpbasepayload.c:
	* gst-libs/gst/video/video-converter.c:
	* gst/encoding/gstencodebasebin.c:
	* gst/playback/gstplaysink.c:
	* gst/playback/gstsubtitleoverlay.c:
	* tests/examples/encoding/gstcapslist.c:
	* tools/gst-device-monitor.c:
	* tools/gst-discoverer.c:
	  common: Stop using GQuark-based GstStructure field name API
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432>

2024-08-09 10:41:57 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioutilsprivate.c:
	* gst-libs/gst/pbutils/gstdiscoverer.c:
	* gst-libs/gst/video/gstvideoutilsprivate.c:
	* gst/playback/gstdecodebin2.c:
	  common: Stop using GQuark-based GstStructure name API
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432>

2024-07-05 19:40:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  gst: structure: Stop using private structure name quark field
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7432>

2024-09-05 22:07:24 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	* gst-libs/gst/video/gstvideoaggregator.c:
	* gst-libs/gst/video/gstvideodecoder.c:
	* gst-libs/gst/video/gstvideoencoder.c:
	  video: Don't overshoot QoS earliest time by a factor of 2
	  By setting the earliest time to timestamp + 2 * diff there would be a difference
	  of 1 * diff between the current clock time and the earliest time the element
	  would let through in the future. If e.g. a frame is arriving 30s late at the
	  sink, then not just all frames up to that point would be dropped but also 30s of
	  frames after the current clock time.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7459>

2024-09-12 10:24:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video-converter.c:
	  video-converter: Add fast path conversions between v210 <-> I420_10 / I422_10
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7504>

2024-09-10 21:25:42 +0200  Carlos Bentzen <cadubentzen@igalia.com>

	* tools/meson.build:
	  meson: gst-play: link to libm
	  gst-play.c depends on libm due to using `round`.
	  Passing by, correct identation of `gst_tools` definition.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7492>

2024-09-07 11:53:05 +0200  Diego Nieto <diego.nieto.m@outlook.com>

	* gst/debugutils/gstfakevideodec.c:
	  fakevideodec: fix minimum allocation pool size
	  Before it was setting the minimum between 2 and the minimum gst_query_parse_nth_allocation_pool(). So, always
	  giving 2 or less.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7465>

2024-08-13 12:07:12 +0900  Chao Guo <chao.guo@nxp.com>

	* ext/gl/gstglimagesink.c:
	* ext/gl/gstglimagesink.h:
	  glimagesink: resize viewport when video size changed in caps
	  When re-negotiation happends and caps is changed, resize the
	  viewport to the corresponding video size in changed caps.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7341>

2024-08-19 10:33:23 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst-libs/gst/gl/meson.build:
	  gl: Fix configure error when libdrm is a subproject
	  When libdrm is a subproject it cannot be used in a configure time check:
	  ERROR: Dependencies must be external dependencies
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7380>

2024-08-20 01:44:57 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/audio/gstaudiocdsrc.c:
	* gst-libs/gst/glib-compat-private.h:
	* gst/audiomixer/gstaudiointerleave.c:
	  gst-plugins-base: use g_sort_array() instead of deprecated g_qsort_with_data()
	  Fixes compiler warnings with the latest GLib versions.
	  See https://gitlab.gnome.org/GNOME/glib/-/merge_requests/4127
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7384>

2024-08-21 16:29:03 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Fix collection identity check
	  Collections can be auto-generated from upstream and yet have exactly the same
	  streams in it.
	  Therefore do a more in-depth check for equality.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3742
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7401>

2024-08-15 17:01:31 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: handle new GstRequestMixMatrix custom upstream event
	  An example use case is the gstwebrtc-api demo, which will cause
	  webrtcsink to forward such events. This lets the end user define a mix
	  matrix without requiring any application code server side.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7363>

2024-08-14 16:09:06 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: fix setting of mix matrix at run time
	  There were two main issues:
	  The mix matrix was not protected with the object lock
	  The code was mistakenly assuming that after updating the mix matrix
	  a reconfigure event sent upstream would be enough to cause upstream to
	  send caps again, and the converter was only reconstructed in ->set_caps.
	  That was not actually enough, as if the new matrix didn't affect the
	  number of input / output channels there was no reason for upstream to do
	  anything after getting the unchanged caps.
	  The fix for this was to have ->transform also recreate the converter
	  when needed, with the added subtlety that depending on the mix matrix
	  the element could be set to passthrough. This means that when setting
	  the mix matrix the converter also had to be recreated immediately to
	  check if the element had to be switched back to non-passthrough.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7363>

2024-08-15 16:32:09 +0200  Mathieu Duponchelle <mathieu@centricular.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: support more numerical types for mix matrix
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7363>

2024-08-19 09:37:51 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Actually drop EOS on old-school pad switch
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7375>

2024-08-16 14:25:53 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin: Don't hold lock when emitting about-to-finish
	  This could trigger actions that re-enter this element
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7368>

2024-07-18 09:26:36 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Remove custom EOS handling
	  Initially introduced in 2017 by 4fcbcf4e4880717fc3cb4099fdfa634c26c5b977 (and
	  follow up commits) and ec7d81f67c844f692113d756db2bd328f92bd554
	  See https://bugzilla.gnome.org/show_bug.cgi?id=773341 and
	  https://bugzilla.gnome.org/show_bug.cgi?id=792693 for more details.
	  This was made to support the legacy behaviour of adaptive demuxers (regarding
	  streams added/removed dynamically).
	  Since then, a lot of things have changed in decodebin3 and related elements
	  regarding how dynamic streams are handled and this custom behaviour is no longer
	  needed.
	  This also removes weird behaviours like EOS being delayed until *all* streams
	  were EOS, which could cause deadlocks downstream.
	  Fixes #3666
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7194>

2024-08-14 10:51:13 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst-libs/gst/pbutils/descriptions.c:
	* tests/check/libs/pbutils.c:
	  pbutils: descriptions: use subsampling factor to get YUV subsampling
	  The algorithm used to determine the YUV subsampling string uses width and height
	  subsampling factor, not the raw subsampling. Otherwise all 4:2:0 YUV frames will
	  be detected as 4:4:4
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7353>

2024-06-21 23:20:48 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Add documentation
	  I didn't find the behavior and purpose of streamsynchronizer documented
	  or intuitive. Eventually I got Edward to explain it to me, which was
	  very helpful. Now I'm contributing some docs so that the next person
	  doesn't have to figure it out by asking around and hoping for an answer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7084>

2024-08-02 15:41:51 +0900  Chao Guo <chao.guo@nxp.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: Add formats supported by #GstGLMemory to raw caps when generating sink pad caps
	  When glupload generates sink caps based on src caps after determining upload method, src
	  caps may only contain RGBA format.
	  In this case, the raw caps on the sink pad generated by glupload will only contain the
	  RGBA format, which will cause caps negotiation fail, because the filter caps used for
	  negotiation by the upstream element may only contain other formats, such as xBGR, etc.
	  Add the formats supported by #GstGLMemory to raw caps to ensure that caps negotiation
	  succeeds.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7061>

2024-08-05 14:37:57 +0200  Piotr Brzeziński <piotr@centricular.com>

	* gst-libs/gst/video/gstvideoencoder.c:
	* gst-libs/gst/video/gstvideoencoder.h:
	  videoencoder: Expose release_frame() and drop_frame() as public API
	  release_frame() can be useful for manually dropping frames without posting QoS messages like finish_frame() would.
	  Matches the same kind of API on the decoder side of things.
	  Modifies the behaviour of release_frame() to make sure events from released frames are stored as 'pending'
	  and pushed before the next non-dropped frame. This is needed because now release_frame() can be called outside of
	  finish_frame(), so we would potentially just lose events and bad things would happen.
	  drop_frame() was also added to match the decoder API. It functions almost identically to finish_frame() without a buffer
	  attached to the frame, except instead of immediately pushing the frame's events, it will store them as pending.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7190>

2024-07-08 11:29:08 +0300  Max Romanov <max.romanov@gmail.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Handle invalid argument properly
	  In case when conn->input_stream is NULL and glib was built with
	  "glib_checks" enabled, g_pollable_input_stream_read_nonblocking()
	  returns -1, but does not set the "err".
	  The call stack:
	  read_bytes() ->
	  fill_bytes() ->
	  fill_raw_bytes()
	  The return value -1 passed up to read_bytes() and incorrectly
	  processed there after "error:" label.
	  This changes the return value to EINVAL.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7210>

2024-07-26 16:10:59 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/gstglframebuffer.c:
	  glframebuffer: Improve error tracing
	  glCheckFrameStatus() can fail by returning 0, and otherwise return a
	  status. Fix the trace to make it clear when we get an unkown status
	  compare to having an error, in which case we also trace the error code.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7291>

2024-07-31 12:35:20 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/xvimage/xvimagepool.c:
	  xv: imagepool: Improve error logging
	  The shm creation function can return a GError, use this to improve the error
	  reporting.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7285>

2024-07-30 16:18:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/xvimage/xvimagepool.c:
	  xvimagesink: Fix crash in pool on error
	  The set_config() virtual function is not support to free the config. As a side
	  effect, when there is protocol error of some sort, we endup with a crash.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7285>

2024-05-29 16:36:29 +0300  Jan Schmidt <jan@centricular.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	  codec-utils: Recognise mjpg mime type
	  Add mjpg to gst_codec_utils_caps_from_mime_codec_single()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6969>

2024-07-29 10:03:58 +0100  Philippe Normand <philn@igalia.com>

	* gst/playback/gstparsebin.c:
	  parsebin: accept-caps handling for elements with unusual pad names
	  In case the last element of the parse chain doesn´t have a sink pad named
	  "sink", send the accept-caps query to the first sink pad of the element.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7255>

2024-07-24 16:52:21 -0400  Nicolas Dufresne <nicolas@tomato.lan>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: Fix syntax error in detiling shader
	  Mesa compiler complains of:
	  error: `rgba' redeclared
	  This regression got introduced by 674e6434289, which introduce
	  colormatrix computation without revoing the pre-declaration of
	  rgba variable.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7231>

2024-07-25 10:34:45 +0800  Shengqi Yu <shengqi.yu@mediatek.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst/videoconvertscale/gstvideoscale.c:
	  videoscale: correct classification error
	  videoscale does not have convert function, so remove the convert
	  description in it's classification. Otherwise, if we want use
	  autovideoconvert to convert colorsapce, autovideoconvert will select
	  videoscale to do convert and this will cause to fail.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7215>

2024-07-19 21:34:31 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/gl/gstglmixerbin.c:
	  glvideomixer: Fix critical when setting start-time-selection
	  It caused a critical, but did not affect functionality because the
	  GValue was passed as-is to the glvideomixerelement which actually does
	  something with the property.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7204>

2024-07-19 20:14:29 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst-libs/gst/gl/meson.build:
	  Revert "meson: Fix invalid include flag in uninstalled gl pc file"
	  This reverts commit 9d719b9937e2ef476cf18f800f54ccac1feea8ee.
	  The old pkg-config implementation errors out if a variable specified
	  in the pkgconfig file is not defined, so this actually broke
	  uninstalled pc files.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7191>

2024-05-28 18:26:56 +0200  Jakub Adam <jakub.adam@collabora.com>

	* gst-libs/gst/video/gstvideoaggregator.c:
	  gstvideoaggregator: preserve features in non-alpha caps
	  Fixes caps negotiation when sink template caps of an element inheriting
	  GstVideoAggregator have features different from the implicit
	  "memory:SystemMemory".
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7102>

2024-05-01 18:05:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	* gst/typefind/gsttypefindfunctionsplugin.c:
	* gst/typefind/gsttypefindfunctionsplugin.h:
	* gst/typefind/gsttypefindfunctionsriff.c:
	* gst/typefind/gsttypefindfunctionsstartwith.c:
	  typefind: Add typefinders for formats that were previously available via ffmpeg
	  Co-Authored-By: Matthew Waters <matthew@centricular.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6505>

2024-07-05 15:09:02 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/plugins/gst_plugins_cache.json:
	  docs: CI insists that this docs update must be applied
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7130>

2024-07-03 18:26:00 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* gst-libs/gst/gl/meson.build:
	  meson: Fix invalid include flag in uninstalled gl pc file
	  ${libdir}/gstreamer-1.0/include is only valid after installation, but
	  extra_cflags are added unconditionally, so we can't use that for
	  include flags.
	  Instead, let's add the include flag via variables, which are different
	  for installed and uninstalled pc files.
	  This is particularly bad for consuming GStreamer via CMake which barfs
	  on non-existent include paths.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7130>

2024-06-26 22:22:47 +0800  He Junyan <junyan.he@intel.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: Add a comment in DMA's propose_allocation query for video meta
	  The video meta API now is a mandatory request for DMA kind negotiation. In
	  current code, the raw method always adds that video meta API in the query
	  of propose_allocation(), so we do not need to do the duplicated task here.
	  Just adding a comment to declare that.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6698>

2024-07-02 15:50:28 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/subparse/gstsubparseelement.c:
	  subparse: remove regex optimized flag explicitly
	  That way the other flags in jit_flags are not touched and
	  flags changes in future only need to be done in one place.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7126>

2024-07-02 15:49:03 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* tests/check/meson.build:
	  gst-plugins-base: put valgrind header availability define into config.h for subparse
	  Make the valgrind header avaibility accessible to any code in
	  gst-plugins-base, currently it was only signalled to unit tests.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7126>

2024-07-02 09:44:25 +0200  Edward Hervey <edward@centricular.com>

	* gst/subparse/gstsubparseelement.c:
	  subparse: Don't use jitted regex when used with valgrind
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7117>

2024-06-29 11:06:07 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/encoding-target.c:
	  encoding-target: Chain up to parent class
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7115>

2024-06-28 17:09:58 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Chain up to parent class finalize
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7115>

2020-04-08 10:40:42 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* meson.build:
	* meson_options.txt:
	  build: Add missing common options that are yielding in subprojects
	  - Align `glib_debug`, `glib_assert` and `glib_checks` options with GLib,
	  otherwise glib subproject won't inherit their value. Previous names
	  and values are preserved using Meson's deprecation mechanism.
	  - Add `extra-checks` and `benchmarks` options in the main project so it
	  can be inherited in GStreamer subprojects.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1165>

2024-06-24 23:30:24 +0900  Seungha Yang <seungha@centricular.com>

	* gst-libs/gst/video/convertframe.c:
	  video: convertframe: Add support for d3d12 conversion
	  Use d3d12convert in case of d3d12 memory
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7095>

2024-06-22 20:56:03 +0300  George Hazan <george.hazan@gmail.com>

	* gst/playback/gstparsebin.c:
	  parsebin: fix compilation problem with Visual C++ 2017
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7089>

2024-06-20 11:37:33 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Fix keyframe drop probe handling
	  We were storing the probe id in a different structure (DecodebinOutputStream)
	  than the pad it is targetting (which is in MultiQueueSlot).
	  The problem is that when re-targetting outputs (to a different slot)... we would
	  end up having an invalid probe id, or not have a reference to an existing one.
	  Instead, store the probe id in the same structure as the pad it's targetting
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7069>

2024-06-20 11:35:08 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Fix detection of selection done
	  We should not assert if there are still some old streams that are waiting to be
	  deactivated.
	  Instead wait for them to be gone before posting the selection done message
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7069>

2024-06-18 16:27:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video-info.c:
	  video-info: Don't crash in gst_video_info_is_equal() if one videoinfo is zero-initialized
	  Instead handle it like gst_audio_info_is_equal() and consider both different.
	  And also add a shortcut for the pointers to both infos being equal.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7055>

2023-09-05 13:04:10 +0200  Alicia Boya García <aboya@igalia.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Add missing INPUT_LOCK() before ensure_input_parsebin()
	  ensure_input_parsebin() has a top comment saying it must be called with
	  INPUT_LOCK taken, but 2 out of 3 usages of the function call it without
	  taking that mutex.
	  This patch adds locking in these two remaining usages.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5279>

2024-06-11 16:42:34 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Don't hold PLAY_ITEMS lock when activating them
	  Once the item is configured it can be activated without holding that lock
	  Fixes #3610
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7015>

2024-06-10 14:56:00 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Always ensure we end up with parsebin or identity
	  This fixes a regression introduced by 6c4f52ea20ce27730501348c6e575b8d35580239
	  There are cases where the input stream will be push-based, time-segment and not
	  have a collection nor caps. This means the event-based checks are not sufficient
	  to decide when/where to plug in a identity or parsebin to process the input.
	  For those corner cases we setup a buffer probe to ensure we always end up with
	  at least a parsebin
	  Fixes #3609
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7010>

2024-05-28 09:24:51 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* gst/subparse/gstsubparseelement.c:
	* tests/check/gst/typefindfunctions.c:
	* tests/files/subrip-short.srt:
	  subparse: fix typefind with small srt files
	  The typefind code was rejecting content smaller than 128 bytes making it
	  impossible to play files with very small srt files.
	  But those can actually be properly detected so fix typefind to allow
	  smaller content and try its best with it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6937>

2024-05-27 14:47:48 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* tests/check/gst/typefindfunctions.c:
	* tests/files/subrip.srt:
	  subparse: add typefind tests
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6937>

2024-05-27 15:56:37 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* gst/subparse/gstsubparseelement.c:
	  subparse: properly group caps
	  No semantic change, but the way caps and macros were grouped was
	  confusing.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6937>

2024-05-28 13:37:39 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* docs/plugins/gst_plugins_cache.json:
	* ext/pango/gstbasetextoverlay.c:
	* ext/pango/gstbasetextoverlay.h:
	  pango: Add a property to compensate for display response time
	  When measuring video latency, one mechanism involves taking a photo
	  with a camera of two screens showing the test video overlayed with
	  timeoverlay or clockoverlay. In these cases, if the display's pixel
	  response time is crappy, you will see ghosting due to which it can be
	  quite difficult to discern what the current timestamp being shown is.
	  This commit adds a property that *also* shows the timestamp in
	  a different (sequentially predictable) location every frame, which
	  makes it easy to tell what the latest rendered timestamp is.
	  For bonus points, you can also use the fade-time of the previous frame
	  to measure with sub-framerate accuracy when the photo was taken, not
	  just clamped to the framerate, giving you a higher precision latency
	  value.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6935>

2024-05-17 14:12:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/rtsp/gstrtspconnection.c:
	  rtspconnection: Use GDateTime instead of gmtime()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6872>

2024-05-28 11:52:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/video.h:
	  video: Document UL_LR / UR_LL video orientation methods correctly
	  They're not flipping along the diagonal axis but a flip and rotation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6936>

2024-05-29 10:31:19 +0200  Edward Hervey <bilboed@bilboed.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Avoid usage of parsebin even more
	  When dealing with push-based inputs, we are now delaying the creation of
	  parsebin/identity until we get all pre-buffer events.
	  We therefore can simplify the handling of new pads being linked and only have to
	  check if upstream can handle pull-based or not.
	  Avoids creating parsebin for parsed upstream data altogether
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6953>

2024-05-29 19:12:21 +0200  Jakub Adam <jakub.adam@collabora.com>

	* ext/gl/gstgldownloadelement.c:
	  gldownload: use gst_gl_sync_meta_wait_cpu()
	  Simple gst_gl_sync_meta_wait() is not sufficient to ensure GL commands
	  are executed before dma-buf devices get to see the buffer.
	  This is the first step that should make the code behave correctly for
	  everybody, although there may be performance penalty. In the future we
	  should introduce a more general sync meta that would allow to move the
	  waiting from gldownload (the producer) to the sink elements (the
	  consumers).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6968>

2024-05-29 14:56:28 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Include the stream-id in ERROR/WARNING/INFO messages
	  Allows application and other users to know to which stream the
	  error/warning/info message originated from.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6964>

2024-05-21 11:32:12 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Specify URI for INFO/WARNING/ERROR messages
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6929>

2024-05-28 04:23:01 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/gl/meson.build:
	* gst-libs/gst/gl/egl/gsteglimage.c:
	* gst-libs/gst/gl/gstglupload.c:
	* gst-libs/gst/gl/meson.build:
	  gl: Fix libdrm dependency detection and usage
	  drm_fourcc.h should be picked up via the pkgconfig dep, not the system
	  includedir directly. All this allows it to be picked up consistently
	  (via the subproject, for example).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6932>

2024-05-21 10:54:46 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Specify stream-id for missing decoder
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6886>

2024-05-21 10:53:50 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/missing-plugins.c:
	* gst-libs/gst/pbutils/missing-plugins.h:
	* tests/check/libs/pbutils.c:
	  pbutils: Missing plugin messages can contain the stream-id
	  In order to help users and applications, allow setting the stream-id for which
	  there is a missing plugin.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6886>

2024-05-24 17:32:21 +0200  Jakub Adam <jakub.adam@collabora.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: update existing sync meta if outbuf has one
	  Instead of always adding a new one, which means the buffer could end up
	  with multiple sync meta instances.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6942>

2024-05-28 11:37:40 +0300  Francisco Javier Velázquez-García <dr.vg@oss-consulting.as>

	* gst/rawparse/gstrawaudioparse.c:
	  docs: Correct pipeline examples in rawaudioparse
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6933>

2024-05-28 16:04:37 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: Fix handling of ID_ODD_SIZE in WavPack typefinder
	  Chunks are always starting on an even position and this flag only
	  specifies that the last byte of the chunk is not valid.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3569
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6941>

2024-04-29 11:26:48 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Ensure we get a collection for parsed inputs
	  When we are dealing with parsed inputs (i.e. using identity), we need to ensure
	  that we have a valid stream collection (and therefore DBCollection) before
	  anything flows dowsntream.
	  In those cases, we hold onto those events until we get such a collection.
	  Fixes #3356
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-04-08 09:56:21 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: New mechanism for handling collection and selections
	  This commit separates collection and selections into a new separate structure:
	  DecodebinCollection.
	  This provides a much cleaner/saner way of dealing with collections being
	  updated, gapless playback, etc...
	  There is now a list of DecodebinCollection in flight, of which two are special:
	  * input_collection, the currently inputted/merged collection
	  * output_collection, the currently active collection on the output of multiqueue
	  Handling GST_EVENT_SELECT_STREAMS is split, by looking for the collection to
	  which it applies. And the requested streams are stored in it. IIF that
	  collection is output_collection we can do the switch, else it will be updated
	  when it becomes active.
	  Detecting which collection/selection is active is done by looking at the
	  GST_EVENT_STREAM_START on the output of the multiqueue.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-04-08 09:34:22 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: minor refactoring to identify selected stream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-04-08 09:28:46 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Debug line cleanups
	  Use identifiable items in log lines instead of random pointers
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-26 08:34:42 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin: Remove unused includes
	  * config.h is not used, plugin/element is registered in another file
	  * play-enum.h is not used
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-25 15:09:14 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Remove un-needed variable
	  We don't do anything with the unknown streams. Detecting that a list of
	  requested streams don't apply to a given collection should be handled
	  before-hand
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-25 12:05:36 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Remove un-needed variable
	  pending_select_streams was only set just before releasing/taking the selection
	  lock in a single place. That temporary lock release is not needed and therefore
	  the variable isn't needed either
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-25 09:33:11 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Remove active_selection list
	  It's a duplicate of the list of slots which have an output. Use that instead.
	  Also when we fail to (re)configure an output, remove it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-25 08:50:21 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Cache slot stream_id and rename more variables
	  * Move the handling of GST_EVENT_STREAM_START on a slot to a separate function
	  * There was a lot of usage of `gst_stream_get_stream_id()` for the slot
	  active_stream. Cache that instead of constantly querying it.
	  * Rename the variables in `handle_stream_switch()` to be clearer
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-24 12:51:43 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Refactor slot/output (re)configuration
	  * Re-use existing function where possible
	  * Only set/reset keyframe probe at unique places
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-24 12:05:29 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Refactor linking input to slot
	  The same sequence of calls was done when doing that
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-24 11:50:17 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: input_unblock_streams: Clarify variable
	  It's a list of pads, not slots
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-22 12:01:06 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Rename multiqueue related functions
	  To make clear on what they apply
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-22 11:55:40 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Refactor/rename slot/output
	  * Centralize associating an output to a slot in one function, including properly
	  resetting those fields
	  * Rename functions to be more explicit
	  * Move code to "reset" an output stream into a dedicated function (will be used
	  later)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-21 12:17:25 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Refactor removal of slot/output from streaming thread
	  The code was identical in several places
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-21 11:04:55 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: rename/clarify eos and draining usage around multiqueue
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-21 08:52:41 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Document/refactor DecodebinInput handling
	  * Rename the function names to be clearer, with prefixes
	  * Pass the input (or stream) directly where appropriate
	  * Document usage, inputs, ownership
	  * Rename variables for clarity where applicable
	  * Avoid double lock/unlock if callee can handle it directly
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-20 09:07:39 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3-parse.c:
	* gst/playback/gstdecodebin3.c:
	  decodebin3: Move gstdecodebin3-parse.c into gstdecodebin3.c
	  Makes it easier to work with LSP
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-19 11:27:23 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Refactor incoming collection handling
	  Simplify its usage by having it directly create the message if the collection
	  changed. This is what caller were always doing and avoids releasing selection
	  locks yet-another-time
	  Also use it in more places to avoid code repetition
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-14 10:39:31 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Rename variable for clarity
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-14 10:06:54 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Refactor GST_EVENT_SELECT_STREAMS handling
	  * The same code is used for the event, regardless of whether it's coming from
	  via a pad or directly on the element
	  * The pending_select_streams list content was never used, switch it to a boolean
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-03-14 09:53:06 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Don't forward select streams if we are handling it
	  Since the introduction of the "SELECTABLE" query, the usage of selection was
	  clarified. We don't need to forward the GST_EVENT_SELECT_STREAMS at this point.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>

2024-05-24 11:35:00 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>

	* ext/pango/gstbasetextoverlay.c:
	  basetextoverlay: use GST_DEBUG_OBJECT instead of GST_DEBUG
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6918>

2024-05-23 18:42:10 +0900  Hou Qi <qi.hou@nxp.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: traverse all structures in caps when transform_caps
	  This is used to avoid negotiation failure between glcolorconvert
	  and downstream element.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6908>

2024-05-21 14:19:26 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: handle rectangle textures for v210/UYVY/YUY2
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6795>

2024-05-16 16:07:49 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: add support for v210->UYVY/YUY2 (and reverse)
	  Also support YUY2/UYVY->YUY2/UYVY conversion
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6795>

2024-05-16 12:12:43 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: add support for UYVY/YUY2->planar yuv (and reverse)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6795>

2024-05-10 11:44:52 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: add support for v210->planar YUV (and reverse)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6795>

2024-05-03 14:17:40 +1000  Matthew Waters <matthew@centricular.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst-libs/gst/gl/gstglcolorconvert.c:
	* gst-libs/gst/gl/gstglcolorconvert.h:
	* gst-libs/gst/gl/gstglformat.c:
	* gst-libs/gst/gl/gstglmemory.c:
	* gst-libs/gst/gl/gstglmemory.h:
	  glcolorconvert: add support for converting to/from v210
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6795>

2024-05-07 23:04:04 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglmemory.c:
	  glmemory: use the GL stride calculation also for downloading into sysmem
	  Allows writing into sysmem with arbitrary strides.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6795>

2024-05-15 10:58:57 -0700  Joshua Breeden <joshua.breeden@kiswe.com>

	* gst/videotestsrc/gstvideotestsrc.c:
	* gst/videotestsrc/gstvideotestsrc.h:
	  videotestsrc: add mutex around cache buffer to prevent race condition
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6849>

2024-04-29 12:15:20 +0200  Yacine Bandou <yacine.bandou@softathome.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Fix deadlock when streams have been flushed before others start
	  To simplify the description, I'm assuming we only have two streams: video and audio.
	  For the video stream, we have the following events :
	  - STREAM_START => stream->wait set to true
	  - NEW_SEGMENT(1) => blocked waiting in gst_stream_synchronizer_wait
	  - FLUSH_START => unblocked
	  - FLUSH_STOP => stream->wait reset to false
	  - NEW_SEGMENT(2) => not waiting, since stream->wait is false
	  Then for the audio stream, we have the following events :
	  - STREAM_START => stream->wait set to true
	  - NEW_SEGMENT(2) => blocked waiting in gst_stream_synchronizer_wait for ever.
	  Note: The first NEW_SEGMENT event and the FLUSH_START, FLUSH_STOP events of the audio stream
	  are dropped before being received by the streamsynchronizer element, because the decodebin audio pad src
	  is not yet linked to the playsink audio pad sink.
	  To fix this deadlock, we don't reset stream->wait to false in the FLUSH_STOP event when it is not
	  waiting for the EOS of the other streams.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6763>

2024-05-20 18:41:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/subparse/gstsubparse.c:
	  subparse: Don't unref seek event twice if forwarding it directly failed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6882>

2024-05-17 09:32:19 -0400  Thibault Saunier <tsaunier@igalia.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Add a way to specify a factory-name when specifying caps
	  In some cases you want to ensure that a specific element factory is used
	  while requiring some specific caps but this was not possible. You can
	  now do `qtmux:video/x-prores,variant=standard|factory-name=avenc_prores_ks`
	  to ensure that the `avenc_prores_ks` factory is used to produce the
	  'standard' variant of prores video stream.
	  This also enhances a bit the documentation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6875>

2024-05-10 10:10:59 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Split up event handler code
	  No changes to behaviour, just split up the big parts into dedicated function for
	  readability
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6825>

2024-05-15 22:17:44 +0900  Seungha Yang <seungha@centricular.com>

	* gst/playback/gstdecodebin3.c:
	* gst/playback/gsturidecodebin3.c:
	* gst/playback/gsturisourcebin.c:
	  decodebin3: Fix caps and stream leaks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6858>

2024-04-26 14:41:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudiobasesink.c:
	* gst-libs/gst/audio/gstaudiobasesrc.c:
	* gst-libs/gst/audio/gstaudioringbuffer.c:
	* gst-libs/gst/audio/gstaudioringbuffer.h:
	  audioringbuffer: Avoid overflows of segment done counter
	  This counter is incremented once for every segment, meaning it would
	  e.g. overflow after 24 days when using 1ms segments. Once that happens,
	  completely wrong positions are reported and invalid memory is handed out
	  for writing/reading the next segments.
	  As the affected variables are unfortunately part of the public API of
	  the struct, a second set of variables is added together with accessor
	  functions and both variables are kept in sync for backwards
	  compatibility.
	  All existing users of the two variables are moved to the new ones but
	  external code might still run into the overflow.
	  This also slightly breaks API as external code updating the variables
	  will have no effect anymore but the only known user of this is
	  pulsesink.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6740>

2024-05-09 12:41:45 +0300  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/audioconvert.c:
	  audioconvert: Add test for 96 channel conversion
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6819>

2024-05-09 09:57:54 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/audio-channel-mixer.c:
	* gst-libs/gst/audio/audio-channels.c:
	  audioconvert: Support converting >64 channels
	  There's nothing requiring <= 64 channels except for getting the reorder
	  map and creating a channel mixing matrix, but those won't be possible to
	  call anyway as channel positions can only express up to 64 channels.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6819>

2024-05-03 22:35:21 -0400  Bill Nottingham <notting@splat.cc>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  pbutils: fix visualization plugins
	  inbuf is persistent, GST_BUFFER_COPY_MEMORY's append behavior is
	  not what we want.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6800>

2024-04-18 17:07:25 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/app/gstappsink.c:
	* gst-libs/gst/app/gstappsrc.c:
	* gst-libs/gst/rtsp/gstrtspconnection.c:
	* gst-libs/gst/video/video-converter.c:
	* gst/compositor/compositor.c:
	* gst/compositor/compositor.h:
	  gst: Move GstQueueArray as GstVecDeque to core
	  And change lengths and indices from guint to gsize for a more correct type.
	  Also deprecate GstQueueArray and implement it in terms of GstVecDeque.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>

2024-04-19 15:44:55 +0200  Rafael Caricio <rafael@caricio.com>

	* gst-libs/gst/pbutils/codec-utils.c:
	* gst-libs/gst/pbutils/codec-utils.h:
	* tests/check/libs/pbutils.c:
	  pbutils: AV1 mime codec
	  Add basic AV1 mime codec param generation based on the spec at https://aomediacodec.github.io/av1-isobmff/#codecsparam
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6696>

2024-05-05 17:48:32 +0100  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: Fix caps negotiation with upstream DMABuf
	  The uninitialized texture target was leading to a NULL texture-target string
	  added to the output caps.
	  Fixes #3519
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6805>

2024-03-05 12:54:32 -0300  Thibault Saunier <tsaunier@igalia.com>

	* ext/gl/gstglvideomixer.c:
	  glvideomixer: Handle pads which have no caps set on mouse events
	  And directly use the pad vinfo instead of getting current caps
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6265>

2024-04-27 00:37:52 +0900  Seungha Yang <seungha@centricular.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst-libs/gst/video/video-converter.c:
	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-format.h:
	* gst-libs/gst/video/video-info.c:
	  video: Add Y216 and Y416 formats
	  The same memory layout as Y212 and Y412 formats, respectively
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6745>

2024-02-23 12:42:56 +0100  Loïc Le Page <llepage@igalia.com>

	* tests/validate/meson.build:
	* tests/validate/videorate/reverse_fast.10_to_1fps.validatetest:
	* tests/validate/videorate/reverse_fast.10_to_1fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_fast.10_to_1fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_fast.10_to_30fps.validatetest:
	* tests/validate/videorate/reverse_fast.10_to_30fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_fast.10_to_30fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_fast.1_to_10fps.validatetest:
	* tests/validate/videorate/reverse_fast.1_to_10fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_fast.1_to_10fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_fast.30fps.validatetest:
	* tests/validate/videorate/reverse_fast.30fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_fast.30fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_fast.variable_to_10fps.validatetest:
	* tests/validate/videorate/reverse_fast.variable_to_10fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_fast.variable_to_10fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_slow.10_to_1fps.validatetest:
	* tests/validate/videorate/reverse_slow.10_to_1fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_slow.10_to_1fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_slow.10_to_30fps.validatetest:
	* tests/validate/videorate/reverse_slow.10_to_30fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_slow.10_to_30fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_slow.1_to_10fps.validatetest:
	* tests/validate/videorate/reverse_slow.1_to_10fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_slow.1_to_10fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_slow.30fps.validatetest:
	* tests/validate/videorate/reverse_slow.30fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_slow.30fps/flow-expectations/log-videorate-src-expected:
	* tests/validate/videorate/reverse_slow.variable_to_10fps.validatetest:
	* tests/validate/videorate/reverse_slow.variable_to_10fps/flow-expectations/log-videorate-sink-expected:
	* tests/validate/videorate/reverse_slow.variable_to_10fps/flow-expectations/log-videorate-src-expected:
	  videorate: add validate tests
	  Add tests with different reverse playback speeds.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>

2024-04-30 11:25:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Fix nullability of (de)serialization functions
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6775>

2024-04-10 15:05:12 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: add support for planar yuv->planar yuv conversions
	  Currently only supported by keeping the same colorimetry and is only
	  a repacking operation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6596>

2024-04-10 14:50:56 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: use color matrices
	  Allows programatically supporting whatever new formats are added to
	  libgstvideo.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6596>

2024-04-25 15:21:20 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/tag/gstexiftag.c:
	  exiftag: Prevent integer overflows and out of bounds reads when handling undefined tags
	  Fixes ZDI-CAN-23896
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3483
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6766>

2024-04-29 15:25:28 +0800  Shengqi Yu <shengqi.yu@mediatek.com>

	* ext/gl/gstgldownloadelement.c:
	* gst-libs/gst/gl/gstglmixer.c:
	  glmixer: Add GL_SYNC_META option to bufferpool
	  when pipline is
	  glvideomixerelement->glcolorconvertelement->gldownloadelement and
	  glcolorconvertelement is not passthrough, the gl bufferpool between
	  glvideomixerelement and glcolorconvertelement will not add gl sync meta
	  during allocating buffer. This will cause that glcolorconvert's inbuf
	  has no sync meta to wait for.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6756>

2024-04-29 16:22:47 +0200  François Laignel <francois@centricular.com>

	* gst-libs/gst/rtp/gstrtcpbuffer.h:
	  rtp: add value annotation for GST_RTP_VALID_VALUE
	  ... otherwise its value is not available to bindings.
	  See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1425#note_2384464
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6762>

2024-04-24 12:42:09 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/tag/gstexiftag.c:
	  exif: add debug category
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6741>

2024-04-26 10:12:23 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstplaybin3.c:
	  playbin3: Handle combiner update in case of errors
	  The assertion that was present before is a bit too harsh, since there is now
	  a (understandable) use-case where this could happen.
	  In gapless use-case, with two files containing the same type (ex:audio). The
	  first one *does* expose a collection with an audio stream, but decoding
	  fails (for whatever reason).
	  That would cause us to have configured a audio combiner, which was never
	  used (i.e. not active).
	  Then the second file plays and we (wrongly) assume it should be activated
	  ... whereas the combiner was indeed present.
	  Demote the assertion to a warning and properly handle it
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3389
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6737>

2024-04-19 11:32:47 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/video/gstvideodecoder.c:
	  videodecoder: Use a frame duration for QoS
	  We prefer to use the frame stop position for checking for QoS since we don't
	  want to drop a frame which is only partially late.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6689>

2024-04-23 07:23:30 +0200  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	* gst-libs/gst/pbutils/encoding-profile.h:
	  encoding-profile: Make (de)serialization functions public
	  This is more convenient and cheaper than going through the `g_value_convert()`
	  hoops
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6712>

2024-04-22 16:05:29 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Remove custom stream-start field if present
	  This field is added by urisourcebin so that we can avoid double-parsing. It's no
	  longer needed after.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6690>

2024-04-19 11:31:54 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturisourcebin.c:
	  urisourcebin2: Adaptive demuxers don't require another parsebin
	  By setting the same field on the GST_EVENT_STREAM_START decodebin3 will be able
	  to avoid plugging in an extra parsebin
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6690>

2024-02-08 20:37:53 +0100  Loïc Le Page <llepage@igalia.com>

	* docs/plugins/gst_plugins_cache.json:
	* gst-libs/gst/audio/audio-channel-mixer.c:
	* gst/audioconvert/gstaudioconvert.c:
	* gst/audioconvert/gstaudioconvert.h:
	* tests/check/elements/audioconvert.c:
	  audioconvert: add possibility to reorder input channels
	  When audioconvert has unpositionned audio channels as input
	  it can now use reordering configurations to automatically
	  position those channels.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5923>

2023-06-19 18:15:49 +0800  Elliot Chen <elliot.chen@nxp.com>

	* gst/subparse/gstsubparse.c:
	  subparse: forward seek event first and return if succeeded
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4889>

2024-03-31 00:30:03 +0900  Seungha Yang <seungha@centricular.com>

	* gst-libs/gst/video/navigation.c:
	* gst-libs/gst/video/navigation.h:
	  navigation: Define mouse double click event
	  Windows and UI toolkits define mouse double click events
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6491>

2024-04-17 09:46:35 +0200  Edward Hervey <edward@centricular.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebin: Fix pad removal for encodebin vs encodebin2
	  Clarify the fact that `encodebasebin->src_pad` is set when using a static source
	  pad (`encodebin`) and when not set it's dynamically added source
	  pads (`encodebin2`).
	  Fixes usage of encodebin2 when profiles are updated
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6667>

2024-04-13 17:50:59 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst-libs/gst/audio/audio-format.c:
	* gst-libs/gst/audio/audio-format.h:
	* gst-libs/gst/video/video-format.c:
	* gst-libs/gst/video/video-format.h:
	* tests/check/elements/videoconvert.c:
	* tests/check/elements/videoscale.c:
	* tests/check/libs/video.c:
	  format: Allow GST_AUDIO/VIDEO_FORMAT_UNKNOWN in _to_string() function
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6630>

2024-04-14 11:55:56 -0400  F. Duncanh <fduncanh@gmail.com>

	* sys/ximage/ximagesink.c:
	  ximagesink: initialize mask for XISelectEvents
	  Fixes #3470
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6631>

2024-04-10 14:58:54 +1000  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: don't overread the end of an array
	  Reverse index iteration needs a - 1 for the initial value.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6595>

2024-04-03 20:35:06 +1100  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: remove unused swizzling using "rgba" chars
	  We use indices now instead.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6595>

2024-02-07 22:13:37 +0530  Sanchayan Maity <sanchayan@asymptotic.io>

	* gst-libs/gst/video/video-converter.c:
	* tests/check/libs/video.c:
	  video-converter: Fix set config not having effect after start
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6594>

2024-04-10 07:22:36 +0200  Edward Hervey <edward@centricular.com>

	* gst/playback/gstparsebin.c:
	  parsebin: Ensure non-time subtitle streams get "parsed"
	  Since https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153 ,
	  subtitle "decoders" (i.e. which decode to raw text) are no longer auto-plugged
	  by parsebin.
	  But if a given format does not have a parser at all, we would end up outputting
	  non-time/non-parsed outputs.
	  In order to mitigate the issue, until such parsers are available, we check if
	  the subtitle stream is in TIME format or not (i.e. whether it comes from a
	  parser or demuxer). If not, we attempt to plug in a subtitle "decoder".
	  Fixes #3463
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6592>

2023-11-23 12:38:30 +0100  Håvard Graff <havard.graff@gmail.com>

	* tests/examples/gl/cocoa/meson.build:
	* tests/examples/gl/cocoa/multiple-glsinks-start-stop.m:
	  examples/gl/cocoa: add an example of a failing scenario with glimagesink
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5708>

2024-04-01 14:31:49 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: Handle WavPack block sizes > 131072
	  These are valid nowadays.
	  Also handle ID_ODD_SIZE and ID_WVX_NEW_BITSTREAM. The parser already
	  handles the former but not the latter.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3440
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>

2024-04-02 16:53:26 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/descriptions.c:
	  pbutils: descriptions: Don't warn on MPEG-1 audio caps without layer field
	  While this is not ideal and won't give too accurate codec descriptions,
	  it is what tsdemux produces.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6511>

2024-02-21 13:56:20 +0100  Robert Rosengren <robertr@axis.com>

	* gst-libs/gst/audio/gstaudioringbuffer.c:
	  audioringbuffer: set waiting to 0 when signaling
	  Reset the waiting thread counter in all places to be consistent
	  when sending signal for the audio ring buffer. This fix applies it to
	  pause, stop and release, which are states that will go into a callback
	  of the subclass. Having the waiting counter reset will avoid having
	  executing thread of the same subclass trying to take the mutex when
	  callong gst_audio_ring_buffer_advance.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6195>

2024-04-02 12:35:37 +1100  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/cocoa/gstglcaopengllayer.m:
	  glcaopengllayer: NULL some fields when freed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6504>

2024-04-02 12:34:16 +1100  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/cocoa/gstglwindow_cocoa.m:
	  glwindow/cocoa: keep a window reference across an async callback
	  Esnures that the window is alive when the callback is fired.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6504>

2024-04-02 12:27:13 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gl/gstglimagesink.c:
	  glimagesink: avoid a possible critical on shutdown
	  It is possible that the close callback can be called after glimagesink
	  is changing state to NULL.  Protect against that by taking the glimagesink
	  lock and some NULL checking.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6504>

2024-04-02 12:25:08 +1100  Matthew Waters <matthew@centricular.com>

	* ext/gl/gstglimagesink.c:
	  glimagesink: unref the potential last ref outside of the glimagesink lock
	  Avoids a deadlock between the state change removing the last ref and
	  the destructer calling the window's on_close handler and trying to
	  take the glimagesink lock.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6504>

2024-04-02 12:22:33 +1100  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/gstglbufferpool.c:
	  glbufferpool: protect release_buffer from multiple concurrent access
	  If two different threads attempt to release buffers at the same time, then the
	  keep-alive-slightly-longer GQueue may become corrupted.  Guard against that with
	  some locking.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6504>

2024-04-02 12:17:50 +1100  Matthew Waters <matthew@centricular.com>

	* gst-libs/gst/gl/cocoa/gstglcontext_cocoa.m:
	  gl/context/cocoa: ensure pixel format lives as long as the context
	  Under some circumstances, the CGLPixelFormatObj was being destroyed too
	  early which could lead to potential use-after-frees.
	  Fix by returning a reference when asked for the pixel format.
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3154
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6504>

2024-04-01 18:00:53 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* gst/videorate/gstvideorate.c:
	* tests/check/elements/videorate.c:
	  videorate: Reset last_ts when a new segment is received
	  This fix all buffers being droped when a new segment is received and
	  average-period property is set.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6502>

2024-03-28 21:46:11 +0100  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* gst-libs/gst/video/navigation.c:
	  navigation: add missing sanity checks
	  On practice a failure happened due to a race condition, the instance
	  already have been freed, but it could also happen if the instance
	  would be null.
	  Instead of crashing this sanity check is a more suitable option,
	  since with G_DEBUG=fatal-warnings it will crash too.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6475>

2024-04-01 14:26:11 +0900  Chao Guo <chao.guo@nxp.com>

	* gst-libs/gst/gl/gstglcolorconvert.c:
	  glcolorconvert: Ensure glcolorconvert does not miss supported RGB formats
	  Add RGB16 and other RGB formats to rgb_formats to ensure glcolorconvert
	  does not miss the RGB formats it supports
	  Signed-off-by: Chao Guo <chao.guo@nxp.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6497>

2024-03-29 12:30:28 +0900  Hou Qi <qi.hou@nxp.com>

	* gst/encoding/gstencodebasebin.c:
	  encodebin: Add the parser before timestamper to tosync list
	  Also need to sync the state of the parser before timestamper with
	  parent.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6481>

2024-03-13 12:49:41 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/audio/gstaudiobasesink.c:
	  Revert "audiobasesink: Don't wait on gap events"
	  This reverts commit 8e923a8e2d7d9391554371045cfe4d0e2d8d0c7a.
	  This caused regressions, see #3303.
	  Without this commit, osxaudiosrc ! osxaudiosink won't work
	  right, but since that hasn't really been a huge problem
	  for years it's probably best to revert this until a proper
	  solution can be figured out.
	  (cherry picked from commit f04f86f3ee4f9eccfd72f473b8ff848161576a90)
	  (cherry picked from commit 93255efece907fdec28366aea6f474af051a1ed0)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6405>

2024-03-19 09:32:39 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Don't hold lock when posting messages or signals
	  There's a very good chance that the receiver might react on those synchronously
	  and call back into uridecodebin3 (ex: for setting the next URI).
	  Make sure we release the lock if we need to do that.
	  Fixes #3400
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6402>

2024-03-18 08:47:18 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gsturidecodebin3.c:
	  uridecodebin3: Handle potential double redirection errors
	  Some elements (like qtdemux) might post a redirection error message twice. We
	  only want to handle it once.
	  Fixes #3390
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6389>

2024-03-14 09:27:10 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Post error messages if there are no streams to output
	  This could happen because:
	  * No streams were selected
	  * Or we end up with no stream selected
	  Also post a warning message if we are missing plugins but there are other
	  streams to output
	  Fixes #3360
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6323>

2024-03-12 16:36:36 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Remove failing stream from active selection also
	  It gets added in get_output_slot()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6323>

2024-03-11 13:27:09 +0100  Thomas Klausner <wiz@gatalith.at>

	* gst-libs/gst/allocators/gstshmallocator.c:
	  shmallocator: fix build on Illumos
	  Closes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3370
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6321>

2023-11-29 14:44:37 +0100  Michiel Westerbeek <happylinks@gmail.com>

	* gst/videoconvertscale/gstvideoconvertscale.c:
	  gstcudaconvertscale, gstvavpp, videoconvertscale: downgrade 'Can't keep DAR' to debug
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5730>

2024-03-11 11:56:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/gl/gstglupload.c:
	  glupload: Do not propose allocators with sysmem
	  None of the GL allocators actually offer a generic alloc() implementation. As a
	  side effect, they cannot be offered as they don't work with generic video
	  buffer pool.
	  Our specialized buffer pool can be dropped by tee or alphacombine as sharing the
	  same buffer pool over two branch is not supported by the pool API.
	  Fixes #3372
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6327>

2024-03-11 18:30:08 +0100  Piotr Brzeziński <piotr@centricular.com>

	* gst-libs/gst/pbutils/gstaudiovisualizer.c:
	  audiovisualizer: Don't wrap temporary memory in buffers
	  Avoids potentially ending up with the buffermemory pointing to already-freed or reused addresses.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>

2024-03-11 13:14:51 +0100  Piotr Brzeziński <piotr@centricular.com>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Avoid wrapping temporarily mapped memory with a GstBuffer and passing that to subclass
	  Memory from gst_adapter_map() could live shorter than the GstMemory that the GstBuffer wraps around it, which in lucky
	  cases 'just' caused a re-use of the same memory for multiple (potentially still in use!) input buffers, but could easily
	  end up pointing to an already-freed memory.
	  Manifested when an AudioToolbox encoder kept getting silence inserted in seemingly random circumstances, turned out
	  to be the memory being re-used by GStreamer at the same time that the AT API was processing it...
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>

2024-03-11 15:58:53 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstplaybin3.c:
	  playbin3: Remove un-needed URI NULL check
	  This will mimic the playbin2 behaviour, which sets the "next" entry to be
	  NULL.
	  The biggest impact this has is that when going back to READY the current play
	  entry will be discarded (instead of being kept around for when you go back to
	  PAUSED/PLAYING).
	  Fixes #3371
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6324>

2024-03-06 09:48:43 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Handle race switching on pending streams
	  find_slot_for_stream_id() will return a slot which has the request stream-id as
	  active_stream *or* pending_stream (i.e. the slot on which that stream is
	  currently being outputted or will be outputted).
	  When figuring out which slot to use (if any) we want to consider stream-id
	  which *will* appear on a given slot which isn't outputting anything yet the same
	  way as if we didn't find a slot yet.
	  Fixes races when doing intensive state changes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>

2024-03-04 15:59:14 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Clear select streams seqnum when resetting
	  At this point there's definitely no pending select streams
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>

2024-03-01 17:38:46 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Only post collection message on actual updates
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>

2024-03-01 17:37:30 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Clear the global collection when resetting
	  This avoids having stray collections when re-using decodebin3
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>

2024-03-11 09:16:01 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstdecodebin3.c:
	  decodebin3: Provide clear error message if no decoders present
	  If we don't do this we will end up with a more cryptic error message (not-linked
	  error from some upstream component).
	  Fixes #3198
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6317>

2024-02-23 15:20:41 +0100  François Laignel <francois@centricular.com>

	* gst-libs/gst/sdp/gstsdpmessage.c:
	  sdp: accept empty attribute value represented as a NULL pointer
	  Some empty media attribute values are set to an empty string, others as a NULL
	  pointer. It seems that code is able to deal with both, except for the UTF8
	  validation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>

2024-02-22 16:36:40 +0100  Piotr Brzeziński <piotr@centricular.com>

	* gst-libs/gst/gl/cocoa/gstglwindow_cocoa.m:
	  macos: Fix glimagesink not respecting preferred size
	  Cocoa version of glwindow only checks the preferred size upon window creation. glimagesink sets the size right before
	  calling gst_gl_window_show(), which might be way after the window is created in some cases. If the size was set too
	  late, glimagesink on macOS would remain 320x240 unless manually resized.
	  This change makes sure to resize the existing window when _show() is called.
	  Curiously, this has always been an issue, but went from manifesting every once in a while to being almost completely
	  broken once old event loop workarounds were removed and gst_macos_main() was introduced.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6185>

2024-03-05 18:37:36 +0800  Xi Ruoyao <xry111@xry111.site>

	* tests/check/meson.build:
	  gst-plugins-base: meson: Fix the condition to skip theoradec test
	  Due to operator priority "not a and b" is interpreted "(not a) and b".
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6260>

2024-03-05 12:58:57 +0000  Tim-Philipp Müller <tim@centricular.com>

	* README.md:
	* RELEASE:
	* meson.build:
	  Back to development
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6261>

=== release 1.24.0 ===

