=== release 1.24.12 ===

2025-01-29 20:12:29 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.24.12

2025-01-28 15:08:03 +0100  Alexander Slobodeniuk <aslobodeniuk@fluendo.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: fix critical warnings on negotiation error
	  This pipeline fails to negotiate on my PC:
	  gst-launch-1.0 v4l2src ! h264parse ! qtmux ! filesink location=t.mp4
	  When it happens some critical glib warnings are emitted:
	  -------------------------------
	  GStreamer-CRITICAL **: 15:09:03.485: gst_mini_object_copy: assertion 'mini_object != NULL' failed
	  GStreamer-CRITICAL **: 15:09:03.485: gst_mini_object_unref: assertion 'mini_object != NULL' failed
	  GStreamer-CRITICAL **: 15:09:03.485: gst_caps_get_structure: assertion 'GST_IS_CAPS (caps)' failed
	  GStreamer-CRITICAL **: 15:09:03.485: gst_structure_set_value: assertion 'structure != NULL' failed
	  GStreamer-CRITICAL **: 15:09:03.485: gst_mini_object_unref: assertion 'mini_object != NULL' failed
	  --------------------------------
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8377>

2025-01-04 10:03:42 +0100  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix wrong usage of GstClockTime vs GstClockTimeDiff
	  This could potentially have caused issues (because of the rest of the code using
	  checks for signed invalid values on a unsigned value)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8293>

2025-01-04 10:03:12 +0100  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Add missing break
	  This would cause the reconfigure path to be called
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8293>

2025-01-04 09:59:55 +0100  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux.c:
	  adaptivedemux2: Add missing break
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8293>

2025-01-04 09:59:34 +0100  Edward Hervey <edward@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux-stream.c:
	* ext/adaptivedemux2/gstadaptivedemux-track.c:
	* ext/adaptivedemux2/hls/gsthlsdemux-stream.c:
	* ext/adaptivedemux2/hls/gsthlsdemux-util.c:
	* ext/adaptivedemux2/hls/gsthlsdemux.c:
	* ext/adaptivedemux2/hls/m3u8.c:
	  adaptivedemux2: Fix usage of GstClockTime vs GstClockTimeDiff
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8293>

2025-01-02 12:24:03 +0100  Jochen Henneberg <jochen@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Undef helper macros after use
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8276>

2024-12-18 08:44:30 +0100  Jochen Henneberg <jochen@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Validate matrix before doing simplified multiply
	  The matrix multiplication makes some assumption about the element
	  values to simplify the math with fixpoint values. If this is allowed
	  for the given matrices is now checked first.
	  Then the debug output for matrix and a comment have been fixed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8276>

2024-12-17 10:48:45 +0100  Jochen Henneberg <jochen@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fixup for orientation matrix parsing
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8276>

2024-12-10 21:34:48 +0100  Jochen Henneberg <jochen@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Use mvhd transform matrix and support for flipping
	  The mvhd matrix is now combined with the tkhd matrix. The combined
	  matrix is then checked if it matches one of the standard values for
	  GST_TAG_IMAGE_ORIENTATION.
	  This check now includes matrices with flipping.
	  Fixes #4064
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8276>

2025-01-07 09:31:26 +0100  Edward Hervey <edward@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Ensure only a single stream-start event is pushed
	  Since we are simulating a single output, we want to ensure only a single
	  stream-start is pushed downstream. We do *not* want to send a (potentially) new
	  stream start event after flushing (like after seeks).
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4146
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8284>

2025-01-09 17:12:54 +0000  Will Miller <will.miller@pexip.com>

	* gst/rtp/gstrtpvp9pay.c:
	* tests/check/elements/rtpvp9.c:
	  rtpvp9pay: fix profile parsing
	  Incorrect parsing of these bits meant that we were incorrectly parsing
	  the VP9 uncompressed bitstream header for some profiles, as the header
	  is of variable length and format depending on the profile. Amongst
	  various unintended effects, this caused the width and height from the SS
	  to be incorrectly parsed and set in the caps.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8283>

2024-11-08 12:06:28 +0100  Piotr Brzeziński <piotr@centricular.com>

	* sys/osxaudio/gstosxaudiosrc.c:
	* sys/osxaudio/gstosxcoreaudio.h:
	  osxaudiosrc: Work around timestamps on iOS not starting from 0
	  On macOS, you always get your own 'timeline' for the AudioUnit session, so timestamps start from 0.
	  On iOS however, AudioUnit seems to give you a 'shared' timeline so timestamps start at a later, non-0 point in time.
	  Simply offsetting seems to do the trick.
	  This was causing osxaudiosrc to not output any sound on iOS.
	  Regressed in 2df9283d3f2ea06af5ebd6db03a6d545cac52f19
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8263>

2024-11-08 12:02:23 +0100  Piotr Brzeziński <piotr@centricular.com>

	* sys/osxaudio/gstosxcoreaudiocommon.c:
	  osxaudiosrc: Fix render callback removal when pausing/stopping
	  At least on iOS, the 'input' callback kept being called after going to PAUSED.
	  Specifying the right type (like in gst_core_audio_io_proc_start()) fixes that.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8263>

2024-11-08 11:57:49 +0100  Piotr Brzeziński <piotr@centricular.com>

	* sys/osxaudio/gstosxcoreaudiocommon.c:
	  osxaudio: Fix AudioOutputUnitStart() deadlock on iOS >=17
	  At some point in iOS 17, this call started waiting for the first render callback (io_proc) to finish.
	  In our case, that callback also takes the ringbuf object lock by calling gst_audio_ring_buffer_set_timestamp(),
	  which results in a deadlock.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8263>

2025-01-06 20:11:58 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.24.11

=== release 1.24.11 ===

2025-01-06 19:48:08 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.24.11

2025-01-06 20:30:51 +0800  Dean Zhang (张安迪) <dean.zhang@mediatek.com>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: release decode only frame in input list
	  For some frames with decode-only flag, the v4l2 decoder will not
	  put them in output list. The corresponding decode-only frames will
	  be still kept in input list, which may cause potential performance
	  issue when the input list is full. So release the decode-only frames
	  according to the decode-only flag after they are processed by decoder
	  driver.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8247>

2024-12-31 11:49:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	* tests/check/elements/matroskamux.c:
	  matroskamux: Consider audio buffers as keyframes when writing out simpleblocks
	  Otherwise mpv complains and considers the file broken.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4142
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8246>

2024-12-31 11:42:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Fix audio-only stream conditions
	  The num_a_streams and related counters are used for pad numbers and don't give
	  the absolute number of streams in this run of the muxer.
	  Also, consider the output audio-only if there are more than 1 audio stream too.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4142
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8246>

2024-12-17 20:08:24 +0100  Christian Meissl <meissl.christian@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix accumulated base offset in segment seeks
	  analog to fix for matroska-demux
	  commit f3c126d07c8a85e76bf5abdfa7f140bbf20545ea
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8244>

2024-12-21 14:39:58 +0100  Robert Mader <robert.mader@collabora.com>

	* docs/gst_plugins_cache.json:
	* sys/v4l2/gstv4l2object.c:
	  v4l2: object: Add P010 format
	  For 10bit content. Tested with HEVC on a Pixel3a (qcom).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8197>

2024-12-12 12:05:04 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtp/gstrtppassthroughpay.c:
	  rtppassthroughpay: ensure buffer is writable before mapping writable
	  It is entirely possible that the incoming buffer into _chain() is not writable
	  and will result in a critical when trying to map().
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8137>

2024-12-04 16:10:46 +0000  Philippe Normand <philn@igalia.com>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  rtpsession: Fix twcc stats structure leaks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8115>

2024-12-03 23:39:54 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.24.10

=== release 1.24.10 ===

2024-12-03 23:29:07 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.24.10

2024-09-27 00:31:36 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Add size check for parsing SMI / SEQH atom
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-244
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3853
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-26 19:16:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check for invalid atom length when extracting Closed Caption data
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-243
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3849
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-27 10:39:30 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Actually handle errors returns from various functions instead of ignoring them
	  Ignoring them might cause the element to continue as if all is fine despite the
	  internal state being inconsistent. This can lead to all kinds of follow-up
	  issues, including memory safety issues.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-245
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3847
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-27 10:38:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Make sure there are enough offsets to read when parsing samples
	  While this specific case is also caught when initializing co_chunk, the error
	  is ignored in various places and calling into the function would lead to out of
	  bounds reads if the error message doesn't cause the pipeline to be shut down
	  fast enough.
	  To avoid this, no matter what, make sure enough offsets are available when
	  parsing them. While this is potentially slower, the same is already done in the
	  non-chunks_are_samples case.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-245
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3847
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-27 09:47:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix error handling when parsing cenc sample groups fails
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-238, GHSL-2024-239, GHSL-2024-240
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3846
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-27 00:12:57 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix length checks and offsets in stsd entry parsing
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-242
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3845
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-26 14:17:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Make sure enough data is available before reading wave header node
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-236
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3843
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-26 09:20:28 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Make sure only an even number of bytes is processed when handling CEA608 data
	  An odd number of bytes would lead to out of bound reads and writes, and doesn't
	  make any sense as CEA608 comes in byte pairs.
	  Strip off any leftover bytes and assume everything before that is valid.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-195
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3841
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-27 15:50:54 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check sizes of stsc/stco/stts before trying to merge entries
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-246
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3854
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-26 18:41:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux_dump.c:
	  qtdemux: Don't iterate over all trun entries if none of the flags are set
	  Nothing would be printed anyway.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-26 18:40:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix debug output during trun parsing
	  Various integers are unsigned so print them as such. Also print the actual
	  allocation size if allocation fails, not only parts of it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-09-26 18:39:37 +0300  Antonio Morales <antonio-morales@github.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix integer overflow when allocating the samples table for fragmented MP4
	  This can lead to out of bounds writes and NULL pointer dereferences.
	  Fixes GHSL-2024-094, GHSL-2024-237, GHSL-2024-241
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3839
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8060>

2024-10-09 11:52:52 -0400  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Put a copy of the codec data into the A_MS/ACM caps
	  The original codec data buffer is owned by matroskademux and does not
	  necessarily live as long as the caps.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-280
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3894
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8058>

2024-09-30 19:19:42 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-ids.c:
	  matroskademux: Skip over zero-sized Xiph stream headers
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-251
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3867
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8058>

2024-09-30 19:06:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Skip over laces directly when postprocessing the frame fails
	  Otherwise NULL buffers might be handled afterwards.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-249
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3865
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8058>

2024-09-30 19:04:51 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't take data out of an empty adapter when processing WavPack frames
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-249
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3865
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8058>

2024-09-30 18:25:53 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Check for big enough WavPack codec private data before accessing it
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-250
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3866
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8058>

2024-09-30 16:33:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Fix off-by-one when parsing multi-channel WavPack
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8058>

2024-09-30 16:32:48 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Only unmap GstMapInfo in WavPack header extraction error paths if previously mapped
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-197
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3863
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8058>

2024-10-04 14:04:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/avi/gstavisubtitle.c:
	  avisubtitle: Fix size checks and avoid overflows when checking sizes
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-262
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3890
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8055>

2024-10-04 13:51:00 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Check size before reading ds64 chunk
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-261
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3889
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8054>

2024-10-04 13:27:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix clipping of size to the file size
	  The size does not include the 8 bytes tag and length, so an additional 8 bytes
	  must be removed here. 8 bytes are always available at this point because
	  otherwise the parsing of the tag and length right above would've failed.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-260
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3888
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8054>

2024-10-04 13:22:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Check that at least 32 bytes are available before parsing smpl chunks
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-259
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3887
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8054>

2024-10-04 13:21:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Check that at least 4 bytes are available before parsing cue chunks
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8054>

2024-10-04 13:15:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Fix parsing of acid chunk
	  Simply casting the bytes to a struct can lead to crashes because of unaligned
	  reads, and is also missing the endianness swapping that is necessary on big
	  endian architectures.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8054>

2024-10-04 13:09:43 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Make sure enough data for the tag list tag is available before parsing
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-258
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3886
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8054>

2024-10-04 13:00:57 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: Check for short reads when parsing headers in pull mode
	  And also return the actual flow return to the caller instead of always returning
	  GST_FLOW_ERROR.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-258, GHSL-2024-260
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3886
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3888
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8054>

2024-10-02 14:44:21 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
	  gdkpixbufdec: Check if initializing the video info actually succeeded
	  Otherwise a 0-byte buffer would be allocated, which gives NULL memory when
	  mapped.
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-118
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3876
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8053>

2024-09-30 16:22:19 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/jpeg/gstjpegdec.c:
	  jpegdec: Directly error out on negotiation failures
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-247
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3862
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8052>

2024-09-26 22:16:06 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Avoid integer overflow when parsing Theora extension
	  Thanks to Antonio Morales for finding and reporting the issue.
	  Fixes GHSL-2024-166
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3851
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8044>

2024-11-29 13:41:54 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/level/gstlevel.c:
	  level: Fix integer overflow when filling LevelMeta
	  The level in GstAudioLevelMeta is represented as a signed 8bit value from 0 to
	  127 (with 127 meaning silence). When converting from double, make sure to clip
	  the value, this also prevent integer overflow in the conversion. This fixes an
	  issue where a lower then -127db is reported and random level with near silent
	  streams (due to integer overflow).
	  Fixes #4068
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8029>

2024-11-28 12:56:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/shout2/gstshout2.c:
	  shout2send: Unref event at the end of the event function
	  The function takes ownership of it and should get rid of it at the end.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8006>

2024-04-03 15:01:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Optionally timestamp RTP packets with their receive times in TCP/HTTP mode
	  Until https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509
	  this was accidentally done inside rtpjitterbuffer for many years, and
	  doing so potentially solves problems on some streams while introducing
	  problems on others.
	  Make this configurable on rtspsrc and default to not set timestamps.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8004>

2024-11-27 12:16:37 +0100  Jonas Rebmann <jre@pengutronix.de>

	* sys/v4l2/gstv4l2videodec.c:
	  v4l2videodec: fix freeze race condition
	  This fixes a possible deadlock between gst_v4l2_video_dec_change_state
	  and gst_v4l2_video_dec_loop on the buffer pool.
	  When stopping capture, the flushing state of the v4l2 capture buffer
	  pool gets reverted in the processing loop after it was set via
	  gst_v4l2_object_unlock (self->v4l2capture) (in
	  gst_v4l2_video_dec_change_state). As a result, gst_v4l2_video_dec_loop
	  does not return and consequently, gst_pad_stop_task gets stuck waiting
	  for the GST_PAD_STREAM_LOCK. To circumvent this, skip acquiring the
	  buffer pool if stopping capture.
	  Suggested-by: Nicolas Dufresne <nicolas.dufresne@collabora.com>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7987>

2024-11-22 18:59:53 +1100  Matthew Waters <matthew@centricular.com>

	* ext/qt/gstqsgmaterial.cc:
	* ext/qt6/gstqsg6material.cc:
	  qt(6)/material: ensure that we always update the context in setBuffer()
	  Scenario is that there are two (or more) GstGLContext's wrapping Qt's GL
	  context from either multiple qml(6)glsink or qml(6)glsrc elements.  Call flow is this:
	  1. material 1 setBuffer()
	  2. material 1 bind()
	  3. material 2 setBuffer()
	  4. material 2 bind()
	  If the call to setBuffer() reuses the same buffer as previous call, then the
	  qt context is not updated in the material.  If however the previously used qt
	  context by the material had been deactivated or freed, then bind() would fail
	  and could result in a critical like so:
	  gst_gl_context_thread_add: assertion 'context->priv->active_thread == g_thread_self ()' failed
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7978>

2024-11-25 15:47:22 +0100  Tomáš Polomský <1155369-polomsky@users.noreply.gitlab.freedesktop.org>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fixed incorrect maximum value for int range
	  There are objects where maximum is not multiplication of the step,
	  e.g. there was a combination where max was 65535 with step 2.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7961>

2024-11-25 14:25:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	* gst/multifile/gstsplitmuxpartreader.h:
	  splitmuxsrc: Convert part reader to a bin with a non-async bus
	  A pipeline always has an async bus, which involves allocating an fd pair. As
	  splitmuxsrc only uses the bus' sync handler, this is not required and can easily
	  cause splitmuxsrc to exceed the fd limit for no good reason.
	  The other features of GstPipeline are also not needed here, e.g. clock selection.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7957>

2024-11-15 11:44:17 +0100  Albert Sjolund <alberts@axis.com>

	* gst/rtpmanager/gstrtphdrext-twcc.c:
	  rtpmanager: don't map READWRITE in twcc header ext
	  There is no need to map the buffer as writable, as there is
	  only a read performed on the mapped buffer. This is in line
	  with other header extensions, as no other extensions maps
	  it as readwrite.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7910>

2020-09-21 16:48:38 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Mux timestampless buffers immediately
	  Instead of leaving them queued indefinitely, or until we're timing out
	  and it's the only buffer queued.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7902>

2024-11-12 11:25:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Don't time out in live mode if no timestamped next buffer is available
	  But also don't wait for a buffer on both pads, which might take forever in case
	  of gaps in one of the streams.
	  The muxer can only advance the time if it has a timestamped buffer that can be
	  output, otherwise it will just busy-wait and use up a lot of CPU.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7902>

2024-11-13 15:49:57 +0100  Robert Rosengren <robertr@axis.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: protect cancellable from unlock/unlock_stop race
	  Protect cancellable from simultaneous unlock and unlock_stop calls from
	  basesrc class.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7897>

2024-11-04 10:49:25 +0100  Stefan Riedmüller <s.riedmueller@phytec.de>

	* docs/gst_plugins_cache.json:
	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Remove little endian marker on 8 bit bayer format names
	  There is no point in having an endian marker on 8 bit bayer format names since
	  it is just one byte. Thus remove it.
	  This also fixes an incompatibility with plugins bad where there is no endian
	  marker on 8 bit bayer format names as well.
	  Fixes: #3729
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7845>

2024-11-05 11:49:32 +0000  Philippe Normand <philn@igalia.com>

	* gst/rtpmanager/gstrtpfunnel.c:
	* tests/check/elements/rtpfunnel.c:
	  rtpfunnel: Ensure segment events are forwarded after flushs
	  gst_rtp_funnel_forward_segment() returns early when the current_pad is set.
	  Without clearing current_pad a critical warning would be emitted when
	  attempting to chain a buffer following a flush.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7835>

2024-11-03 17:36:46 +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/ky.po:
	* po/lt.po:
	* po/lv.po:
	* po/mt.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  gst-plugins-good: update translations
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7820>

2024-10-31 17:46:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph265depay.c:
	  rtph264depay, rtph265depay: various parameter-set string handling fixes
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7812>

2024-10-30 20:40:12 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.24.9

=== release 1.24.9 ===

2024-10-30 20:33:30 +0000  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.24.9

2024-10-29 17:39:02 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Consider timestamps before segment start to map to segment start
	  Instead of mapping them to running time 0, which is wrong if e.g. the segment
	  base is not equal to 0.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7798>

2024-10-29 15:30:59 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Use first running time on the initial header instead of 0
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7798>

2024-08-08 12:28:11 +0000  Johan Sternerup <johast@axis.com>

	* gst/rtpmanager/rtptwcc.c:
	* tests/check/elements/rtpsession.c:
	  twcc: Handle wrapping of reference time
	  Previously the wrapping of the 24-bit reference time was not handled
	  correctly when transforming it into GstClockTime. Given the unit of 64ms
	  the span that could be represented by 24 bits is 12 days and depending
	  on the start value we could get a wrapping problem anytime within this
	  time frame. This turned out to be particularly problematic for the GCC
	  algorithm in gst-plugins-rs which tried to evict old packages based on
	  the "oldest" timestamp, which due to wrapping problems could be in the
	  future. Thus, the container managing the packets could grow without
	  limits for a long time thereby creating both CPU and memory problems.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7792>

2024-10-29 16:43:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtppassthroughpay.c:
	  rtppassthrough: fix rtp-stats message compatibility with GstRTPBasePayload
	  "clock-rate" and "pt" are G_TYPE_UINT in the base class, so let's
	  keep them like that here too, since the entire purposes of the
	  passthrough element is to fake being a payloader. The types in the
	  message don't have to be consistent with the types in the caps.
	  Reverts part of commit a6fa53b7 of !7526
	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7552#note_2576653
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7784>

2024-10-25 12:02:54 +0200  Ognyan Tonchev <ognyan@axis.com>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsource.c:
	  rtpmanager: skip RTPSources which are not ready in the RTCP generation
	  If a stream has an 'irregular' frame rate (e.g. metadata) RTCP SR
	  may be generated way too early, before the RTPSource has received
	  the first packet after Latency was configured in the pipeline.
	  We skip such RTPSources in the RTCP generation.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7777>

2024-10-03 12:48:31 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix parsing of matrix with 180 rotation
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7665>

2024-09-26 09:15:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Check fourcc of a second CEA608 atom instead of assuming it's cdt2
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7583>

2024-09-23 16:48:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* docs/gst_plugins_cache.json:
	  doc: good: Update documentation cache
	  video4linux2 plugin now maps RGB15 which his didn't before.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-09-18 13:14:32 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Fix a gvalue leak on error
	  In case we failed enumerating the supported interlacing mode, we leaked the
	  gvalue.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-09-17 14:27:46 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2videodec.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: dec/enc: Flag leaked caps
	  We never free class held template caps, so flag the one that wasn't already
	  flagged.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-08-15 16:54:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: object: Fix condition check to emit error
	  The check was reversed, so we could only emit a pipeline error
	  if there was no element associated with the object.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-09-17 13:28:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Always tell capture queue that we want to set the CSC
	  Not all drivers supports it, but in general we want to try and match the
	  negotiated caps, so lets always try to set the CSC.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-08-15 16:01:03 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: object: Fix support for format:Interlaced in caps probe
	  This notably follow the way we order the template and keeps the
	  format:Interlaced caps at the end. This change also fixes
	  an early skip check, that would skip if a driver only supports
	  alternate interlacing for a specific format. It also fixes
	  a bug where only the last resolution of a discrete frame size
	  was allowed to use format:Interlaced. Finally, similar to template
	  caps code, simplify the caps for earch featurs, making the debug output
	  manageable and (marginally) improve negotiation speed.
	  This change will make it easier to introduce memory:DMABuf.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-08-15 13:07:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2: Move M2M template caps probe into v4l2object
	  This allow reusing the code that produces output and capture devices
	  templates. This fixes the lack of Interlaced caps feature for M2M
	  devices such as decoder, encoder or converters.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-08-14 15:26:58 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: object: Remove over indentation
	  This is a style fix, no functional changes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-08-14 15:21:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: object: Map GST/V4L2 formats in a C array
	  This makes it easier to add new format in the future without
	  forgetting to update one of the numerous switch case. This
	  will also help mapping DRM formats.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-08-14 10:13:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	  v4l2object: Expose convertion from v4l2 fourcc to GstVideoFormat
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-08-14 09:52:54 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: Change dimensions format desc field to flag
	  The boolean naming wasn't obvious, and having this as a flag makes
	  the structure a little more compact.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7563>

2024-07-29 09:07:40 +0800  Shengqi Yu <shengqi.yu@mediatek.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2object: append non colorimetry structure to probed caps
	  If the stream has a special colorimetry that is not in the colorimetry
	  list, it will cause negotiation to fail. We should allow passing any
	  colorimetry, so add an extra structure without the colorimetry field.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7570>

2024-09-24 09:50:34 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Skip zero-sized boxes instead of stopping to look at further boxes
	  A zero-sized box is not really a problem and can be skipped to look at any
	  possibly following ones.
	  BMD ATEM devices specifically write a zero-sized bmdc box in the sample
	  description, followed by the avcC box in case of h264. Previously the avcC box
	  would simply not be read at all and the file would be unplayable.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7565>

2024-08-13 15:07:07 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* ext/qt6/qt6glwindow.cc:
	* ext/qt6/qt6glwindow.h:
	  qml6glsrc: Reduce capture delay
	  In qml6glsrc, we capture the application by copying the back buffer into
	  our own FBO. The afterRendering() signal is too soon as from the apitrace, the
	  application has been rendered into a QT internal buffer, to be used as a cache
	  for refresh.
	  Use afterFrameEnd() signal instead. This works with no delay on GLES. With GL
	  it seems to reduce from 2 to 1 frame delay (this may be platform specific). A
	  different recording technique would need to be used to completely remove this
	  delay.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7562>

2024-09-18 12:34:39 +0200  Piotr Brzeziński <piotr@centricular.com>

	* docs/gst_plugins_cache.json:
	* gst/rtp/gstrtppassthroughpay.c:
	* gst/rtp/gstrtppassthroughpay.h:
	  rtppassthroughpay: Fix reading clock-rate and payload type from caps
	  They were using wrong types - while uint is correct technically, for compatibility reasons caps have them as signed int.
	  Values are now correctly read + added simple guards just to be sure.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7552>

2024-09-19 12:12:53 +0200  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development after 1.24.8
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7549>

=== release 1.24.8 ===

2024-09-19 12:01:21 +0200  Tim-Philipp Müller <tim@centricular.com>

	* NEWS:
	* RELEASE:
	* gst-plugins-good.doap:
	* meson.build:
	  Release 1.24.8

2024-08-13 16:38:37 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2videoenc.c:
	  v4l2: encoder: Add dynamic framerate support
	  This is not trully supported in V4L2, but we can emulate this similar to
	  what other elements do. In this patch we ensure that 0/1 is supported by
	  encoders (caps query),and uses a default of 30fps whenever we need to
	  set a framerate into the driver.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7545>

2024-09-11 13:23:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Include end padding in the block duration for Opus streams
	  It has to be included in the block duration but in GStreamer we're not
	  including it in the buffer duration, so it has to be added again here.
	  Not including it in the block duration can lead to fatal errors when playing
	  back with Firefox if there are more padding samples than actual samples, e.g.
	  > D/MediaDemuxer WebMDemuxer[7f6a0808b900] ::GetNextPacket: Padding frames larger
	  > than packet size, flagging the packet for error (padding: {13500000,1000000000},
	  > duration: {6000,1000000}, already processed: false)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7517>

2024-09-05 22:07:24 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/adaptivedemux2/gstadaptivedemux.c:
	* gst/deinterlace/gstdeinterlace.c:
	* gst/monoscope/gstmonoscope.c:
	* gst/shapewipe/gstshapewipe.c:
	* gst/videomixer/videomixer2.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/7518>

2024-09-11 08:28:39 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
