The LCEVC_ReceiveDecoderBase() and LCEVC_ReceiveDecoderPicture() APIs from the
LCEVCdec SDK can return LCEVC_Error if the enhancement data is wrong. This
change improves the lcevcdec element to check for those errors and stop the
pipeline when that happens.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9057>
The profile argument passed to gst_encode_base_bin_set_profile is now
transfer-full. This issue was noticed after commit
6beb709d43d2023e7e5dc8f1ee1323bc28c9d1d8 which fixed profile refcount handling
in transcodebin.
Driving-by, an encoding profile leak was also fixed in _set_profile, in case
it's called for an already active element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9167>
When GST_DISABLE_GST_DEBUG is defined, the log context functions were
missing their stub implementations, causing link errors. Add the missing
stub functions to the disabled debug section.
Also add (nullable) annotation to gst_log_context_get_category since
it can return NULL when debugging is disabled.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9122>
Use gst_pad_push_event() instead of manually getting peer pad and calling
gst_pad_send_event() otherwise the pad probe on the sinkpad wouldn't get the
events.
Also ensure navigation events are posted to the bus even when gst_pad_push_event()
fails in the render function, so navigation remains functional regardless
of rendering issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9130>
compositor will record rendering commands using multiple threads
(i.e., blending commands are recoded using thread pool, and
background one is recorded on aggregate thread).
And there can be temporary refcount increase (so not writable).
Updates fence once all rendering commands have been submitted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9157>
Adding an option to keep them no matter what.
Log files are often pretty large and keeping them around can be annoying,
usually people won't look at logs files for passing tests, and we do not
even print them out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9127>
If the driver has size alignment requirement, suggest the
difference between aligned size required by diver and the actual
size of the frame as padding requirement to the upstream element.
This will ensure that the buffer size allocation is as per the
driver requirement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8971>
When using the custom WebKitMediaSrc element (used by WebKit and able to
perform an initial seek in playbin), a stall caused by streamsynchronizer
was detected during an initial seek. The flow of events revealed that the
intertwining of the initial configuration of the streams with the reset
caused by the flush events from the seek left streamsynchronizer in an
inconsistent state:
streamsynchronizer0:sink_0 (video) events, starting before the seek:
stream-start --> Sets the stream to wait
flush-stop --> Clears the stream wait flag
caps
tag
segment
stream-collection
(buffers start to come and flow properly)
streamsynchronizer0:sink_1 (audio) events, happening after seek:
(no flush events, because the stream hadn't been initialized when the seek happened)
stream-start --> Sets the stream to wait
caps
segment
(stalled because the stream is in wait mode!)
The code in streamsynchronizer expects that all the streams are in wait
state before releasing all of them at once. The flush on the video stream
broke that assumption and that's why the audio stream is never released in
that scenario.
Avoiding the clearing of the wait flag on flush-stop isn't an actual solution
to the problem, as it creates other side effects and at least makes the
gst-editing-services/seek_with_stop test to timeout. The alternate solution
implemented in this patch consists on analyzing if the other streams different
from the one newly added (after the flush) aren't waiting (which would mean
that they've all been unlocked after all of them were waiting before), and,
in that case, mark the new stream as also not waiting.
A new test_stream_start_wait test case has been added to demonstrate this
problem. The test case creates a video stream, pushes a buffer, then
simulates a seek by pushing flush-start, flush-stop, stream-start and segment
events. Note that the flush-stop clears the video stream waiting flag.
After that, a new audio stream is created and stream-start and new segment
events are sent. Note that stream-start will set the audio stream to wait.
Then a buffer is pushed on each stream. In the failing case, the test hangs.
In the working case (after this fix), the test runs properly because the
fact of having seen a stream-start also helps to clear the wait flag.
A second new test_stream_start_wait_sparse test has also been added to prove
that this mechanism can also work with sparse streams (a special case of the
current stream-start handling code). This test behaves like the previous one,
but there's no video buffer after the seek (it'll come in the future, as the
stream is sparse, but actually never comes). The buffer after the seek in the
audio stream starts at its due time. Streamsynchronizer is able to ignore
the wait for the video stream and produce audio buffers on time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4544>
Nowadays VK_EXT_debug_report is considered deprecated and it's recommended to
replace it it VK_EXT_debug_utils, which offer a way to ignore messages
considered false positives.
The approach is to try the extension first, if available at compilation time, if
not or if it fails to load, VK_EXT_debug_report fallbacks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9119>
Convert buffer metadata getter/setter methods to Python properties for
more intuitive access. This makes the API more Pythonic by allowing
direct property assignment (buf.pts = 42) instead of method calls
(buf.set_pts(42)).
Updated properties: flags, pts, dts, duration, offset, offset_end
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9134>
