=== 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 ===
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