There are a bunch of plugins that you need for webrtc support, and
it's not obvious at all to users which those are.
With this commit, srtp, sctp and dtls options will be auto-enabled if
the webrtc option is enabled.
Requires meson 1.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5505>
The v4l2codecs H.265 decoder uses the
GstH265SliceHdr::entry_point_offset_minus1 array so make sure that it is not
freed before decoding the frame.
Before this patch, some H.265 input would segfault in
gst_v4l2_codec_h265_dec_fill_slice_params() when executing the line:
guint32 entry_point_offset = slice_hdr->entry_point_offset_minus1[i] + 1;
Make sure that the array is not freed before using it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5499>
While adding arbitrary tile support, a round up operation was badly
converter. This caused the Y component of the stride to be 0. This
eventually lead to a crash in glupoad preceded by the following
assertion.
gst_gl_buffer_allocation_params_new: assertion 'alloc_size > 0' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5458>
The caps that were sent by the caps event can be retrieved from the sinkpad
using gst_pad_get_current_caps(). This is more reliable than using cur_caps as
we know exactly which caps upstream selected when the UVC host didn't select a
format, yet.
This further allows to simplify the check, if the uvcsink has to wait for the
caps event before switching to the internal v4l2sink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The probe passes all events except the EVENT_CAPS. Installing and removing the
probe doesn't provide any additional value.
Install an event function and always handle EVENT_CAPS. Use the caps_changed
field, to decide, if the element has to do anything special on a EVENT_CAPS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
Move the sanity checks to the beginning of the function. Make the actual effect
of the function more obvious and reset the flags in the end.
This should make it easier to understand what this function is doing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The probe that installs the buffer probe is already on the correct pad. There is
no need for a separate function to install the probe.
While at it, change the signature of the probe functions to GstPadProbeCallback
to avoid the cast when installing the probes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The uvcsink calculates the caps for the format that the UVC host selected. The
gst_uvc_sink_parse_cur_caps() sets these caps as cur_caps as a side effect. This
behavior is surprising as cur_caps is later updated to reflect the actually used
caps.
Just return the configured caps to avoid side effects. This makes the function
easier to understand. Update the function name to reflect the new behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
The only job of the event peer probe is to catch the upcoming caps event
and be able to react with the sink change. All other events that are
passing the pad shall be passed and ignored.
Since the probe is a blocking probe, there is no use in returning
with GST_PAD_PROBE_OK on other events. Otherwise the event would just
be blocked.
Since we are handling the probe removal of the probe already in the
event switch, we can remove the second explicit probe removal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4994>
This commit ports functionality from the `rtpsrc` to make the `ristsrc`
work with dynamic payload types.
It adds two properties:
- `caps`
- `encoding-name`
These can be used to make the `ristsrc` receive other payload types than
the MPEG TS one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5422>
Since DXVA does not support some profiles such as HEVC RExt,
vendor specific decoding API is still required.
When decoder is negotiated with d3d11 caps, decoder will convert
semi-planar frame to planar since semi-planar format (e.g.,
DXGI_FORMAT_NV12) is not supported by CUDA/D3D11 interop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5409>
Use gst_codec_utils_caps_get_mime_codec() in pbutils for codec
strings. That function gives more elaborate RFC 6381 compatible
strings than the helper functions in gstmdphelper.c, such as
"avc1.F4000D".
Remove the helper functions, as they were only used from dashsink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5404>
The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5324>
Moves outputting frames to a task on the source pad, bringing vtdec in line with vtenc.
This brings possible performance improvements thanks to decoupling queueing new frames from outputting processed ones.
The queue length is limited to `2*DBP` to prevent decoding too far ahead compared to what we're pushing downstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5163>
This was easy to trigger when testing with e.g. vtenc ! vtdec ! glimagesink and closing the sink via window button,
causing GST_FLOW_ERROR to be received by the output loop, stopping it with the queue still full. This made the
enqueue_buffer() callback to lock waiting for space in our queue, while handle_frame() was waiting for the internal
VideoToolbox queue to free up, so that VTCompressionSessionEncodeFrame could finish. As the output loop was not
running, both functions waited forever.
Fixed by 1) immediately emptying our queue when GST_FLOW_ERROR is received (like we already did with _FLUSHING)
and 2) unconditionally setting the flushing flag in finish_encoding() when it sees the output loop stopped because
of GST_FLOW_ERROR, so that enqueue_buffer() will immediately discard any new frames coming out of VideoToolbox.
Both of those make sure we never run into the both-queues-full scenario.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5303>
As a short-term solution before full d3d12 rendering feature,
copy decoded d3d12 texture to shared d3d11 texture in order to use
existing various d3d11 implementations such as conversion, resizing,
and videosink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5356>
There is no need to use DRM dumb pool if buffer to
render is already a DMABuf, just import it and render it.
This fixes a DMAbuf memory leakage when waylandsink downstream
element exports DMABuf while waylandsink is configured to be
DMABuf exporter (drm-device=/drv/dri/card0):
gst-launch-1.0 v4l2src io-mode=4 ! gtkwaylandsink drm-device=/dev/dri/card0
leakage identfied with command:
watch "cat /sys/kernel/debug/dma_buf/bufinfo | grep attached "
Fixes#2729
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5350>
There is no need to use DRM dumb pool if buffer to
render is already a DMABuf, just import it and render it.
This fixes a DMAbuf memory leakage when waylandsink downstream
element exports DMABuf while waylandsink is configured to be
DMABuf exporter (drm-device=/drv/dri/card0):
gst-launch-1.0 v4l2src io-mode=4 ! waylandsink drm-device=/dev/dri/card0
leakage identfied with command:
watch "cat /sys/kernel/debug/dma_buf/bufinfo | grep attached "
Fixes#2729
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5350>
Don't update info's size with the VA image reported data size for single plane
images, since drivers might allocate bigger space than the strictly required to
store the image, but when we dump the buffer as is (using filesink, for example)
the produced stream is corrupted. For multi-plane images video meta is required
to read/write them.
We updated info's size because gstreamer-vaapi did it too, but the reason to
update it there was for uploading and rendering surfaces (commit c698a015).
Furthermore, this patch adds an error message if the allocated data size for the
image by the driver is lesser than the expected because it would be a buggy
driver.
Fixes: #2959
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5308>
Even if decoder is negotiated with CUDA memory feature, if downstream
proposed no buffer pool, assume that the pool size is unknown.
And disable zero-copy if there's no more free output surface.
Or, in case of reverse playback, always copy frames.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5338>
Even if the segmentation feature value is not updated,
the parsed "segmentation_update_map" and "segmentation_temporal_update"
values should not be cleared as it's referenced during lower
level bitstream parsing. Also, don't use assert() in parser
unless it's clearly impossible condition.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5334>
If DPB is full already, GstH265Decoder::new_picture() might fail if
subclass uses fixed size picture pool and its size is equal to the DPB
size. Call the new_picture() after DPB is cleared in gst_h265_decoder_dpb_init()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5333>
Issue is that when amc was producing a codec-data buffer, a
GstVideoCodecFrame was being popped off the internal queue. This meant
that the codec-data was being associated with the first input frame and
the second (first encoded buffer) output buffer with the second input
frame. At the end (assuming one input produces one output which seems
to hold in my testing and how the encoder is currently implemented)
there would be an input frame missing and would be pushed without any
timing information. This would lead to e.g. muxers rejecting the buffer
without PTS and failing to mux.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5330>
This is consistent with the librtmp-based old rtmp plugin and ffmpeg.
While some servers require a valid flash-version, others are failing
with a too long or any flash-version at all.
By changing to the same default as in the old plugin and in ffmpeg,
GStreamer will at least behave the same and will work and fail with the
same servers without setting a flash-version.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5293>
It is similar to NV12 but has 10bits per channel instead of 8.
As it is supported by many modern GPUs, VA-API and an increasing
number of Wayland compositors, let's support it as well.
Also bump the required libdrm version accordingly and add a temporary
define for the WL_SHM format.
Tested with Weston, Mutter and Sway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5275>
Old versions of mesa doesn't support VASurfaceAttribDRMFormatModifiers. To
solve it, by just ignoring the modifiers assuming that linear is accepted and
produced, the creation of frames will be tried again without that attribute.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5256>
This patch removes the code duplication of input buffer importation, in all the
va elements that import video frames. It defines a synthetic object whose
members are required to create a new input buffer and do the importation of the
upstream buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5257>
Setting the surface source rectangle has been omitted so far. As a side effect
surface created with padded width/height are being scaled down. Fix this using
the viewporter source rectangle configuration. This can later be enhanced
to support crop meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5259>
When we consider the DMA kind caps as input, the input_state->info
only contains the video format of GST_VIDEO_FORMAT_DMA_DRM, which
is not enough for va plugins. The new info in base encoder contains
the correct video info after the DMA caps parsing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5189>
Since d3d11convert and its variant elements does not enable basetransform's
passthrough, passthrough allocation query needs to be handled
manually in order to respect downstream element's min/max buffer
requirement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5255>
* Library versioning should not be used for plugins since it will add
-{version}.dll suffix (and versioned libraries on Linux with symlink).
Then the library file name and plugin init function name mismatch
will result in blacklisted plugin.
* Don't define BUILDING_GST_CODECS, makes no sense
* Don't define G_LOG_DOMAIN, which should be used only for libraries,
not plugins
* Depends on gstcodecparsers libary, not gstcodecs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5249>
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.
By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:
http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc
This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
Adding cudaipc{src,sink} element for CUDA IPC support.
Implementation note:
* For the communication between end points, Win32 named-pipe
and unix domain socket will be used on Windows and Linux respectively.
* cudaipcsink behaves as a server, and all GPU resources will be owned by
the server process and exported for other processes, then cudaipcsrc
(client) will import each exported handle.
* User can select IPC mode via "ipc-mode" property of cudaipcsink.
There are two IPC mode, one is "legacy" which uses legacy CUDA IPC
method and the other is "mmap" which uses CUDA virtual memory API
with OS's resource handle sharing method such as DuplicateHandle()
on Windows. The "mmap" mode might be better than "legacy" in terms
of stability since it relies on OS's resource management but
it would consume more GPU memory than "legacy" mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4510>
If glyphrun unit is changed in a single line, there could be
overlapped background area which result in drawing background
twice. Adding geometry combine so that background geometry objects
with the same color can be merged and rendered at once
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5179>
Latest MSYS2 MinGW provides these now, so we don't need to define them
if they're already present in the header.
The AudioClient3 GUID requires the Windows 10 SDK, so it's only
available in the latest MinGW, and the MinGW in Cerbero is too old.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5155>
VA decoders implementation has been verified from 1.18 through 1.22
development cycles and also via the Fluster test framework. Similar
to other cases, we can prefer hardware over software in most cases.
At the same time, GStreamer-VAAPI decoders are demoted to NONE to
avoid collisions. The first step to their deprecation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2312>
These 10bit formats are identical to NV12_16L32S, but 64bytes of data is being
prefixed with 16bytes data with four pixels of lower 2bits per byte. For
MT2110T, the lower two bits set so each bytes contains a column of 4 pixels,
also describe as tiled lower 2 bits. MT2110T has been chosen as a name to match
the vendor chosen name. This format is unlikely to exist for other vendors.
For MT2110R, the 2 low bits are in raster order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3444>
There is currently no way for applications to know if the stream has
been properly terminated by the server or if the network connection
was disconnected as EOS is sent in both cases.
Adding a property so connection errors can be reported as errors
allowing applications to distinguish between both scenarios.
Fix#2828
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5115>
Pass GstVideoInfoDmaDrm or GstVideoInfo whenever possible, avoiding passing
strange combination of GstVieoFormat + modifier. Even though we don't have any
at the moment, this also allow supporting GstVideoFormat that are not supported
in our DRM integration.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5120>
According to libva API description, cu_qp_delta in VAConfigAttribValEncHEVCFeatures
is supposed to be used as a flag not the value of depth. And if flag enabled,
diff_cu_qp_delta_depth should be decided by log2_diff_max_min_luma_coding_block_size.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5068>
Rework the va_map_unlocked() after we keep mapping behavior (whether to
use derive) consistent with allocator_try stage. Also remove the flag
for iHD case because pitch/stride difference between vaCreateImage and
vaDeriveImage only possibly happen on iHD by now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5046>
In gst_va_allocator_try, the first try is to use derive_image, if it
succeeds, we should use info from derived image to create bufferpool.
If derive fails, then try create_image and give created image info
to the pool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5046>
vtenc has an async output queue, which we only iterate over after another frame is enqueued.
At the very least it means we're always a frame behind the fastest possible output.
In edge cases it's also bug-prone - for example if we only have 1 frame, the downstream caps negotiation
will never happen.
This commit adds a separate task running on the source pad, which only iterates over the output queue
and pushes frames out as soon as they're put there. The queue length is limited to ensure we don't encode
too far ahead compared to what downstream can consume. Any failures that occur when pushing data downstream
will be signalled in self->downstream_ret so that other parts of code can act accordingly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4967>
Sending an EOS event is actually really bad because rtpbin doesn't
handle that very well. It was only being used as a way to notify
webrtcbin to check if re-negotiation is needed.
We don't need that anymore, since changing the direction is enough to
notify webrtcbin to check for re-negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5045>
lc3enc:
- encodes raw audio into lc3 format
- uses the default bitrate property and frame duration
from the caps to determine the byte count of
the encoded frames if it is not specified in
the downstream caps after negotiation
- uses the same byte count value for all the channels
- all the common session configuration parameters
are passed in the src caps
lc3dec:
- decodes an lc3 encoded audio
- sink caps should contain all the common session configuration
params
- uses frame_duration and frame_bytes (byte count) in the sink
caps as parameters along with sample rate and channel count
- byte count is same for all the channels
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4376>
`srt_rejectreason_str` doesn't give us a unique string for every
possible reason. Peers can define their own reasons and SRT just gives
us the string `"Application-defined rejection reason"` for all of them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4948>
Adding new subtitle overlay element. It's a bin which is wrapping
two internal elements dwritesubtitlemux and dwritetextoverlay.
* dwritesubtitlemux: A new internal element to aggregate subtitle
buffers and to attach the aggregated subtitle buffers on
video buffer as meta.
* dwritetextoverlay: Extracts/renders the subtitle meta and
discard the meta after rendering.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4934>
There's no reason to release GstMemory manually at all.
If we do release GstMemory, corresponding GstBuffer will be
discarded by GstBufferPool baseclass because the size is changed
to zero.
Actual cause of heavy CPU usage in case of fixed-size pool
(i.e., decoder output buffer pool) and if we remove GstMemory from
GstBuffer is that GstBufferPool baseclass is doing busy wait in acquire_buffer()
for some reason. That needs to be investigated though, discarding
and re-alloc every GstBuffer is not ideal already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4935>
The current way of using gst_video_info_set_format() will change all
fields of the GstVideoInfo. We only need to change its format, stride
and offset fields.
In order to keep the consistency with th common drm API, we rename the
gst_va_video_info_from_dma_info() into gst_va_dma_drm_info_to_video_info().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4883>
The current way only selects the best video format from the first
structure of the caps. The caps like:
video/x-raw(memory:VAMemory),drm-format=(string)NV12; \
video/x-raw(memory:VAMemory),format=(string){ NV12, Y210 }
Will just choose NV12 as the result, even the bitstream is 10 bits.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4928>
As VK_FORMAT_FEATURE_2_xxx are defined as static const variable, the
vscoce C compiler prevents the initialization of the vk_usage_map
structure with error "C2099: initializer is not a constant".
Init the structure separately.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4904>
The current way of using parent's copy_metadata() virtual function will
selectively filter out some meta such as crop meta. That virtual function
should be used when copying input buffer's meta data into output buffer,
not suitable when importing the input buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
When the input buffer has crop meta, and we need to do copy, we
should consider the uncropped video size and copy the full size
of video memory.
The video meta in this case should contain the full uncropped
resolution info. We can use it to create full size va buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4887>
Adding Direct3D11 backend Qt6 QML videosink element, qml6d3d11sink.
Implementation details are similar to the qt6 plugin in -good
but there are a few notable differences.
* qml6d3d11sink accepts all GstD3D11 supported video formats (e.g., NV12).
* Scene graph (owned by qml6d3d11sink) will hold dedicated and sharable
RGBA texture which belongs to Qt6's Direct3D11 device, instead of sharing
GStreamer's own texture with Qt6.
* All rendering operations will be done by using GStreamer's Direct3D11 device.
Specifically, upstream texture will be copied (in case of RGBA)
or converted to the above mentioned Qt6's sharable texture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3707>
Now all codec baseclasses can inform subclasses of correct max DPB size,
and exception handling (e.g., emergency bumping in h.264) has been
improved as well. Smaller number of additional DPB frame allocation
seems to be safe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4878>
Adding GST_CUDA_CRITICAL_ERRORS env variable so that program can be
terminated on unrecoverable error.
Example)
GST_CUDA_CRITICAL_ERRORS=2,700 gst-launch-1.0 ...
In this example, CUDA_ERROR_OUT_OF_MEMORY(2) and
CUDA_ERROR_ILLEGAL_ADDRESS(700) are registered as critical error
and program will be aborted on those errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4729>
If two senders use the same multicast IP and port then new_session_pad()
may try to add a srcpad to the same stream twice.
stream->srcpad is updated but gst_element_add_pad() fails the second
time. As a result stream->srcpad points to a deleted object and
access in gst_sdp_demux_stream_free() fails with a segfault.
Just ignore the second pad. Nothing useful can be done with it anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4603>
The `gst_video_decoder_negotiate_pool` function expects the
`decide_allocation` function to always provide a pool and will fail to
negotiate if the pool is missing. If we return immediately (even if we
don't need to do anything special) negotiation will fail if the
downstream element does not propose a pool.
Fix by chaining up to the default `decide_allocation` function which
adds a fallback pool if one was not already proposed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4630>
Adding DirectWrite text rendering elements
* dwriteclockoverlay: Equivalent to clockoverlay
* dwritetimeoverlay: Equivalent to timeoverlay
* dwritetextoverlay: Similar to textoverlay but subtitle is not
supported
Newly added elements support system memory and d3d11 memory
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4826>
This new property allows setting of PES stream number for AAC audio
and AVC video streams.
The stream number is subject to the following constraints:
1. it must be between 0 and 15 for video
2. it must be between 0 and 31 for audio
Currently the PES stream number is hard-coded to zero for these
stream types.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4822>
Add support for 10/12/14/16 bit depths . This consists of multiple parts.
First is the parsing of caps, which pulls out the bitness and endianness
from the video/x-bayer format.
Second, gst_bayer2rgb_split_and_upsample_horiz() is split into two similar
functions, one for 8bit bayer handling and another for 16bit bayer handling.
The content is basically identical, except one uses 8bpp and the other 16bpp
inputs and outputs, and they each use different ORC code to match. The 16bpp
variant also handles endian swapping. There is now a wrapper called
gst_bayer2rgb_split_and_upsample_horiz() which selects the correct function
based on bpp from the parser.
Third, gst_bayer2rgb_process() is extended to handle both 8bit and 16bit
bayer data. Yet again there are matching ORC functions to handle the 16bit
data. This time however the 16bit handling of data is slightly special. The
ORC is not able to emit opcodes for 'x2 mergelq', so the trick here is to
store the BG and GR longs into separate 'dtmp' temporary buffer, and then
do one more ORC post-processing step, compensate for the less-than-16bpp
bitness using left shift, and reorder them into the destination frame
using 'mergelq' .
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr16le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add comments regarding which LINE()s point to which data in the
temporary buffer and a large comment explaining how the buffer
is processed. This will hopefully be useful to someone, as the
code is not obvious. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Instead of passing a single element of GstBayer2RGB structure into the
gst_bayer2rgb_split_and_upsample_horiz(), pass the entire pointer and
let the funciton pick out whatever it needs out of the structure. This
is a preparatory patch. No functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Pass all three parameters used by the LINE() macro to the LINE() macro
and unroll the code for readability. Add more comments regarding which
of these LINE()s point to which data in the temporary buffer to make
the code less confusing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The j variable is used as an iterator further down in this code, but
here it can be just inlined in the macro parameters to make the code
easier to read. This is done in preparation for further changes. No
functional change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
The bayer2rgb process implemented doesn't support in-place tranform.
This element doesn't implement a "transform_ip" vmethod of
GstBaseTransform it will revert to using the "tranform" vmethod.
It's misleading to set it to TRUE, here. Change this to FALSE.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Add support for conversion to 10/12/14/16 bit bayer pattern.
The implementation is rather simplistic, just take the ARGB
input, generate 16-bit data out of it instead of 8-bit, shift
them as required by the output bitness, and apply endian swap.
Example usage:
```
$ gst-launch-1.0 videotestsrc num-buffers=1 ! \
video/x-raw,width=512,height=512,format=ARGB ! \
rgb2bayer ! \
video/x-bayer,format=bggr12le ! \
filesink location=/tmp/bayer12.raw
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
Even if we don't yet know what the echo probe format is, we want to be able to
provide silence for the reverse path, so that when the probe becomes available,
there is no ambiguity around what time period the new set of samples are for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
The probe's info may not precisely match the dsp's info. For instance,
the number of channels or their layout might be different.
```
GStreamer-Audio-CRITICAL **: 16:21:32.899: the GstAudioInfo argument is not equal to the GstAudioMeta's attached info
```
This broke in d5755744c3.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4849>
A race condition can occur in `srtpdec` during the READY -> NULL transition:
an RTCP buffer can make its way to `gst_srtp_dec_chain` while the element is
partially stopped, resulting in the following critical warning:
> Got data flow before segment event
The problematic sequence is the following:
1. An RTCP buffer is being handled by the chain function for the
`rtcp_sinkpad`. Since, this is the first buffer, we try pushing the sticky
events to `rtcp_srcpad`.
2. At the same moment, the element is being transitioned from PAUSED to READY.
3. While checking and pushing the sticky events for `rtcp_srcpad`, we reach the
Segment event. For this, we try to get it from the "otherpad", in this case
`rtp_srcpad`. In the problematic case, `rtp_srcpad` has already been
deactivated so its sticky events have been cleared. We won't be pushing any
Segment event to `rtcp_srcpad`.
4. We return to the chain function for `rtcp_sinkpad` and try pushing the
buffer to `rtcp_srcpad` for which deactivation hasn't started yet, hence the
"Got data flow before segment event".
This commit:
- Adds a boolean return value to `gst_srtp_dec_push_early_events`: in case the
Segment event can't be retrieved, `gst_srtp_dec_chain` can return an error
instead of calling `gst_pad_push`.
- Replaces the obsolete `gst_pad_set_caps` with `gst_pad_push_event`. The
additional preconditions checked by previous function are guaranteed here
since we push a fixed Caps which was built in the same function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4844>
A race condition can occur in `srtpdec` during the READY -> NULL transition:
an RTCP buffer can make its way to `gst_srtp_dec_chain` while the element is
partially stopped, resulting in the following critical warning:
> assertion 'parent->numsinkpads <= 1' failed
This can occur when the first RTCP buffer is received during the READY -> NULL
transition. If deactivation of the `rtp_srcpad` has already reached
`post_activate`, the sticky events are removed from this Pad. In this case,
`gst_srtp_dec_push_early_events` branches to the generation of a stream id
using `gst_pad_create_stream_id`. This function ensures that the element
doesn't own more than 1 sink pad. Since `srtpdec` owns two of them, the
assertion fails.
This commit uses `gst_element_decorate_stream_id` which doesn't perform this
check. The preconditions is not necessary in this particular context since the
stream id for the RTP / RTCP pads are derived from the same id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4844>
Change the internal GstVideoInfo structure in the GstVaDmabufAllocator to
GstVideoInfoDmaDrm in order to keep track of the exported DRM format by the
driver, and thus removing the DRMModifier quark attached as qdata in the
GstMemory. Though, the exposed API isn't updated yet; that has to go in a
second iteration.
Also this patch clean up some code (remove an unused buffer size assignation)
and fix some typos in documentation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4821>
The VA has its internal video format mapping(because different drivers may
have different interpretation for the same format), so we should convert the
info in GstVideoInfoDmaDrm into the according video info based on that mapping.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4821>
Some surface formats such as GST_VIDEO_FORMAT_Y42B and GST_VIDEO_FORMAT_RGB
can be created but can not be exported as DMA buffer. You can not say that
this is a driver bug because the driver may never want to share this kind of
surface out of libva.
And this function will be used to detect modifiers later, so the error message
will be annoying.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4821>
The proxy callback for the notify::last-message was emiting the signal
again on the child, which caused an infinit loop. We could swap the child
and the user data to signal to the bin instead, but it was found that proxying
this signal was not very useful. Typical use case it to set silent=0 and use
deep-notify feature. Proxying that signal just duplicate that output which
isn't very useful.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4766>
Decoder bounded CUDA memory is allocated by driver and the pool size
is fixed. Since we don't know how many buffers would be held by
downstream non-CUDA element, we should download such CUDA memory
and release it back to decoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4810>
Currently the uvcsink is only capable to run in an application
that is handling the state transitions of the pipeline properly
by checking on streaming event from the uvcsink.
This code is improving the element by adding an fakesink to
consume possible videostream flow in case the pipeline state
is not changing on hosts streamoff.
This is helpfull when using local gst-launch pipelines where
the streaming event is not monitored to change the pipelines
state.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1304>
This patch adds an element to stream video data to an uvc video gadget.
The element handles the uvc events STREAMON, STREAMOFF, SETUP and DATA.
to start, stop and configure the video buffer flow by the use of pad
probes. It works with linux kernels of versions higher than v6.1.
The element makes use of the v4l2sink proxy property v4l2sink::device
to locate the corresponding device to parse the configfs for additional
data.
The code in uvc.c is basically derived from /lib/uvc.c in
https://git.ideasonboard.org/uvc-gadget.git.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1304>
Updated API usage appropriately, and now we have a versioned package to
track breaking vs. non-breaking updates.
Deprecates a number of properties (and we have to plug in our own values
for related enums which are now gone):
* echo-suprression-level
* experimental-agc
* extended-filter
* delay-agnostic
* voice-detection-frame-size-ms
* voice-detection-likelihood
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
Allow a project to use gstreamer-full as a static library
and link to create a binary without dependencies.
Introduce the option 'gst-full-target-type' to
select the build type, dynamic(default) or static.
In gstreamer-full/static build configuration gstreamer (gst.c)
needs the symbol gst_init_static_plugins which is defined
in gstreamer-full.
All the tests and examples are linking with gstreamer but the
symbol gst_init_static_plugins is only defined in the gstreamer-full
library. gstreamer-full can not be built first as it needs to know what plugins
will be built.
One option would be to build all the examples and tests after
gstreamer-full as the tools.
Disable tools build in subprojects too as it will be built at the end of
build process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4128>
The `switch (n_rear)` supports up to 5 rear channels, but our channel
set only had space for 3. Size the set properly to fix this.
This didn't actually cause any memory unsafety as `PUSH_CHAN` would stop
incrementing `n_rear` if the channel set is already full.
Thanks to @alatiera for noticing this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4712>
adjust log level from GST_ERROR to GST_WARNING when h264 caps have
codec_data but no avc format or have no codec data or stream-format.
Because theses are not real errors, it is easy to mislead if print error
logs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4675>
The resolution of VP9 video can be changed without keyframe.
The change detected by MSDK/VPL should be negotiated with downstream.
Only the situation can be fixed here if the changed resolution is less than or equal to the initial surface resolution.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4450>
New vulkan formats don't match the number of planes with the number of memories
attached to the buffer. This patch changes the pattern of using planes for
traverse the memories with the number of attached memories.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
It's a generalization of the original gst_vulkan_get_or_create_image_view().
The reason for passing the whole VkImageViewCreateInfo structure rather than
just the missing fields, is because VkImageSubresourceRange and
VkComponentMapping can be different and those are most of VkImageViewCreateInfo.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
This is going to be used when the pool is used by a video decoder for
VK_IMAGE_USAGE_VIDEO_DECODE_DST_BIT_KHR, since the frame allocation needs the
VkVideoProfileInfoKHR, and for that here GstCaps is used to wire it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
The specification says:
VUID-vkAllocateMemory-pAllocateInfo-01713
must pAllocateInfo->allocationSize be less than or equal to
VkPhysicalDeviceMemoryProperties::memoryHeaps[memindex].size where memindex =
VkPhysicalDeviceMemoryProperties::memoryTypes[pAllocateInfo->memoryTypeIndex].heapIndex
as returned by vkGetPhysicalDeviceMemoryProperties for the VkPhysicalDevice that
device was created from.
Though this can be catch by the validation layer, the requested frame size
depends on the use case so it's better to check this restriction by our code.
This patch also makes use of this new function to find memory type index,
and removes the unused function to find memory type index, which, as GstVulkan is
considered unstable, we can do it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
The purpose of this function is to get more info about the mapped Vulkan format
from the GStreamer format, since they can be multiple Vulkan formats for one
GStreamer format.
Also a Vulkan format may have certain usage and aspects that must be verified.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
Originally the opened device only created one queue of one family queue, to say
graphics one. This approach felt short when other queue family is required not
shared with the graphics queue family, for example video decoding.
This new approach proposes to create those queues with supported families. For
now, only video decoding and encoder are created, if they are available.
In order to hold multiple queues opened, an array of VkDeviceQueueCreateInfo is
held along the live the device object, because it's used to traverse or get the
opened queues.
The algorithm to choose which queues create (or open) is to look for the queue
with more family bits, which also supports the one we are requesting, thus
minimizing the number of global queues of a certain family to create.
Nonetheless, the number of queues to open per family is set to be all of them,
widening the possibility of parallelism.
Also, this commit do a cosmetic refactor the assigning the physical device
nearer where it's used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
Also adds a meson option to enable them.
The symbol GST_VULKAN_HAVE_VIDEO_EXTENSIONS is an alias of
defined(VK_VERSION_1_4) || (defined(VK_VERSION_1_3) && VK_HEADER_VERSION >= 238
if the option is allowed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4351>
max-qp and min-qp will set the same quantizer scale for I/P/B frames,
while max-qp-i/p/b and min-qp-i/p/b enable the max/min quantizer for I,P,B
frame separately. When max/min-qp and max/min-qp-i/p/b are given
simultaneously, the later set one will overide the previous one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4589>
when play rtsp stream with playbin3 enabled, there are some critical logs:
g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-video'
g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-audio'
g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-text'
self->collection could be NULL when READY->PAUSED if the pipeline
is live, then it will fallback to query playbin2's property,
we can call gst_play_streams_info_create_from_collection
directly, it will check self->collection internal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4460>
The devices list returned by g_udev_client_query_by_subsystem() may
contain udev devices in disorder path name. For example, on some
platform it may contain renderD129 before renderD128 device. This
will cause we register wrong va plugin name. In this case, the
renderD129 will be registered as default plugins such as vah265dec,
while the renderD128 will be registered as varenderD128h265dec.
This conflicts with the non-udev version of gst_va_device_find_devices().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4643>
When transitioning from state PAUSED to READY, the sctpenc element
could previously be stuck in an endless loop trying to resend data
in case the underlying sctp stream was in the process of
resetting. usrsctp_sendv() would repeatedly return EAGAIN with the
result that 0 bytes were sent and then sctpenc would retry forever.
To bring sctpenc out of the resend loop we just need to inform the
sink pad that it is flushing, which is already done for the associated
data queue, but we also need to set the bools associated with the
sinkpads that are used as the loop criterion.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4601>
This is no longer needed since the introduction of `gst_macos_main()` in 1.22.
Before that existed, we had a patch for GLib in Cerbero, which did work but made it
impossible to update GLib at all. The code being removed was a fail-safe in case of
running without said patch being applied. It's no longer needed, since for macOS
we just wrap our GStreamer with an NSApplication using `gst_macos_main()`.
Warnings will be displayed if no NSApp/NSRunLoop is found wherever needed,
pointing the user towards using the new API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4366>
Depending on the exact output format, 0x00 may be a better default for
padding than 0x80. 0x00 is the recommended padding value when used in
CDP (and cc_data) but is not when used in s334-1a. See CTA-708-E 4.3.5
amd SMPTE 334-1-2007 5.3.2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4578>
A blocking pad probe is added on new sink pads, it's usually removed after the
caps have been negotiated or the signaling state switched to stable, but if that
never happens and the pad is released we kept the pad probe active, leaving the
pad blocked, preventing clean disposal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4529>
Current implementation can in some cases detect
that all data is sent but in reality it is not,
leading to a push to an unlinked pad.
This is a race between the probe used to track data sent and a
call to close.
This patch sends an EOS before starting the close procedure
and then waits for the EOS event to come through to the
src pad before commencing with tear down.
This ensures that any queued data before EOS is flushed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4462>
On MacOS with homebrew, the openssl library is not
properly detected with pkg-config.
So disable the test compilation if openssl
is not properly detected along with libcrypto.
libcrypto is detected but it uses the system one
which leads to the error:
your binary is not an allowed client of /usr/lib/libcrypto.dylib for
architecture x86_64
See more details from Apple:
https://developer.apple.com/forums/thread/124782
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4481>
`webrtc->signaling_state` (from) and `new_signaling_state` (to) had the
same value when printed in a trace log. This commit adds a
`old_signaling_state` variable to hold the previous value, so that the
print statement works as intented.
Spotted by: Mustafa Asım REYHAN
Fixes#1802
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4362>
The addresses we get from `resolve_host_finish()` (via
`resolve_host_async()`, `resolve_host_main_cb()`, `on_resolve_host()`,
`g_resolver_lookup_by_name_finish()`) must be freed. Otherwise we leak
memory.
Leak found and confirmed fixed with GCC AddressSanitizer.
Change-Id: If32d24452d626234f01b253b77a7d6d16eac1cee
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4469>
Similar to cairooverlay element but this element emits "draw"
signal with Direct3D11 render target view, so that an application
can render/overlay/blend on the given render target view
without any copy operation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4415>
We need to create the sink caps and src caps dynamically for different
platforms. By default, the vpp init function create static pad template
and the compatibility and flexibility of the platform are too poor.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4177>
Enable dynamic capability support for msdkav1dec, msdkh264dec,
msdkh265dec, msdkmjpegdec, msdkmpeg2dec, msdkvc1dec, msdkvp8dec,
msdkvp9dec.
The gstmsdkdec elements can create the sink caps and src caps
dynamically for different platforms.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4177>
We need to create the sink caps and src caps dynamically for different
platforms. By default, the dec init function create static pad template
and the compatibility and flexibility of the platform are too poor.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4177>
Enable dynamic capability support for msdkav1enc, msdkh264enc,
msdkh265enc, msdkmjpegenc, msdkvp9enc, msdkmpeg2enc.
The gstmsdkenc elements can create the sink caps and src caps
dynamically for different platforms.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4177>
We need to create the sink caps and src caps dynamically for different
platforms. By default, the enc init function create static pad template
and the compatibility and flexibility of the platform are too poor.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4177>
By default, msdk plugin will register all encoders and decoders
on any platform, even unsupported encoders and decoders will be
registered. Dynamically register encoders and decoders besed on
the supported codecs on different platforms.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4177>
The original code was:
if (!gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {
goto error;
} else {
stream->key = buf;
}
So use "srtp-key" if it is set so a non NULL buffer. The condition was
incorrectly inverted in ad7ffe64a6 to:
if (gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {
stream->key = buf;
} ...
Fix the condition so it works as originally intended and avoid accessing
'buf' uninitialised.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4401>
We should behave similarly to video parsers so we can use:
- accept-template as we can also accept caps with missing fields.
- accept-intersect to do quick check with the pad template caps as it is
enough. Users should have figured the appropriate full caps on a
previous caps query
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4341>
The framerate should only be replaced (and corrected for alternating field)
when it is parsed from the bitstream. Otherwise, the upstream framerate
from caps should be trusted and assumed correct.
Related to gst-plugins-bad!2020
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4259>
One race condition is the fact that the window object
can be destroyed while running some routine in the UI
thread (such as resizing). To avoid that situation we make
UI thread hold a reference on the window object while it's
running.
Other probpematic case is when the window handle is reused:
if we stop and start the pipeline very fast,
so the sink creates a new window object that is going to use
the same window handle as the previous one.
And finally the case when the pipeline is stopped immediatelly
right after starting, this one is also handled in this commit.
NOTE: a unit test that reproduces this cases have been added
in the previous commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4260>
Existing codes rely on modified argc value by g_option_context_parse()
but g_option_context_parse_strv() is used in case of Windows.
Count arguments after the option parsing manually.
Fixing command "gst-inspect-1.0.exe -b"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4313>
