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>
Move the GstStructure field into public struct for direct access, that's
easier than having to call a function to get it. It is not an API/ABI
breakage to extend the public structure of a GstMeta because they are
always allocated by inside GStreamer. The structure is exposed already
by gst_custom_meta_get_structure() which does not return a copy/ref, so
it is locked into holding a GstStructure forever anyway.
Also add gst_meta_register_custom_simple() because most of the time only
a name is required, tags and transform functions are more niche
use-case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5385>
If there are multiple Wayland event listeners in different threads we
get the formats and modifiers pushed concurrently which leads to
segfault from GArray methods. This patch protects the array.
The problem occurs e.g. when using vaapipostproc together with Qt
qmlglsink, QtWayland will get the events as well as VAAPI.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5280>
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>
Right now we split the RTP header from the current buffer into a new
buffer and aggregate those buffers for later processing if the
depayloader creates an output buffer.
This is cumbersome as it happens even if none of the incoming RTP
buffers carries RTP header extensions at all just because header
aggregation has been enabled in the depayloader class.
This commit will start aggregation only in case that there really are
RTP header extensions available on an incoming RTP buffer. The check
is trivial and cheap. Once activated we keep aggregation active for
all buffers. The active state is reset on state change READY_TO_PAUSE.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5278>
scanlines->m1 = same line of the previous field
scanlines->t0 = line above of the current field
scanlines->b0 = line below of the current field
scanlines->mp = same line of the next field
Deinterlacing a field weaved frame:
When deinterlacing the top field, the next bottom field is available
(part of the same frame). but when deinterlacing the bottom field,
the next top field (part of the next frame) is not available and
scanlines->mp equals NULL.
In this case it's better to use greedy algorithm using the prevous field
(twice) rather then linear interpolation of the current field.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5331>
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet
then this can't be used for bitrate estimation, and also not for
constructing the next RTCP SR. Both would end up with completely wrong
values, and an RTCP SR with wrong values can easily break
synchronization in receivers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
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>
