A zero-sized box is not really a problem and can be skipped to look at any
possibly following ones.
BMD ATEM devices specifically write a zero-sized bmdc box in the sample
description, followed by the avcC box in case of h264. Previously the avcC box
would simply not be read at all and the file would be unplayable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7564>
This notably follow the way we order the template and keeps the
format:Interlaced caps at the end. This change also fixes
an early skip check, that would skip if a driver only supports
alternate interlacing for a specific format. It also fixes
a bug where only the last resolution of a discrete frame size
was allowed to use format:Interlaced. Finally, similar to template
caps code, simplify the caps for earch featurs, making the debug output
manageable and (marginally) improve negotiation speed.
This change will make it easier to introduce memory:DMABuf.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7540>
In qml6glsrc, we capture the application by copying the back buffer into
our own FBO. The afterRendering() signal is too soon as from the apitrace, the
application has been rendered into a QT internal buffer, to be used as a cache
for refresh.
Use afterFrameEnd() signal instead. This works with no delay on GLES. With GL
it seems to reduce from 2 to 1 frame delay (this may be platform specific). A
different recording technique would need to be used to completely remove this
delay.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7351>
Timestamps are untouched by default, but the new mode can now be enabled to replace RTP timestamps
with ones generated from the buffer PTS. Making it an enum in case different modes are needed in the future.
That allows for a rtpjitterbuffer to do proper drift compensation, so that the stream coming out of gst-rtsp-server
is not drifting compared to the pipeline clock and also not compared to the RTCP NTP times.
Most of the code is borrowed from rtpbasepayload, as it's exactly its behaviour which I wanted to bring here.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7526>
Valgrind complains about uninitialized memory used in an ioctl
Syscall param ioctl(VKI_V4L2_G_TUNER).reserved points to uninitialised byte(s)
at 0x719294F: ioctl (ioctl.c:36)
by 0x3126A817: gst_v4l2_fill_lists (v4l2_calls.c:185)
by 0x3126A817: gst_v4l2_open (v4l2_calls.c:589)
by 0x3123F1C2: gst_v4l2_device_provider_probe_device (gstv4l2deviceprovider.c:122)
by 0x3123F648: gst_v4l2_device_provider_device_from_udev (gstv4l2deviceprovider.c:301)
by 0x3123F998: provider_thread (gstv4l2deviceprovider.c:395)
by 0x796FA50: ??? (in /usr/lib/x86_64-linux-gnu/libglib-2.0.so.0.7200.4)
by 0x710CAC2: start_thread (pthread_create.c:442)
by 0x719DA03: clone (clone.S:100)
Address 0x44008a34 is on thread 11's stack
in frame #1, created by gst_v4l2_open (v4l2_calls.c:524)
Uninitialised value was created by a stack allocation
at 0x3126A024: gst_v4l2_open (v4l2_calls.c:524)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6144>
This is not trully supported in V4L2, but we can emulate this similar to
what other elements do. In this patch we ensure that 0/1 is supported by
encoders (caps query),and uses a default of 30fps whenever we need to
set a framerate into the driver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7352>
It has to be included in the block duration but in GStreamer we're not
including it in the buffer duration, so it has to be added again here.
Not including it in the block duration can lead to fatal errors when playing
back with Firefox if there are more padding samples than actual samples, e.g.
> D/MediaDemuxer WebMDemuxer[7f6a0808b900] ::GetNextPacket: Padding frames larger
> than packet size, flagging the packet for error (padding: {13500000,1000000000},
> duration: {6000,1000000}, already processed: false)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7502>
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>
splitmuxsink can't possibly know how much latency it will introduce as it always
keeps one GOP around before outputting something. This breaks the latency
configuration of the pipeline and we're better off just pretending that
everything downstream of the sinkpads is not live.
Especially muxers that are based on aggregator and time out on the latency
deadline can easily misbehave otherwise as the deadline will be exceeded usually.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7499>
When no ports are given, gst_jack_get_ports() is called to get all the
(physical) output ports but then the result is ignored, triggering the
"No physical output ports found..." error.
Instead, move the queried ports to the variable we're going to use
later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7474>
If two (or more) rtpfunnel elements are cascaded, then only one will
realistically have information on the particular ssrc that is in use for a
particular input stream. As such, any key unit requests may never reach the
corresponding encoder.
This has been discovered by combining simulcast and BUNDLE with webrtcbin.
simulcast uses one rtpfunnel, and BUNDLE uses another rtpfunnel.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7405>
```
In file included from ../subprojects/gst-plugins-good/ext/qt6/gstqsg6material.cc:31:
../subprojects/gst-plugins-good/ext/qt6/gstqsg6material.h:69:17: error: private
field 'mem_' is not used [-Werror,-Wunused-private-field]
69 | GstMemory * mem_;
| ^
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7414>
This patch addresses the issue where GStreamer would throw an error when
attempting to use bt2100-hlg colorimetry with V4L2, which is not
supported by the current V4L2 kernel. When bt2100-hlg colorimetry is set
from caps, the check for transfer (GST_VIDEO_TRANSFER_ARIB_STD_B67) is
bypassed.
The main improvement is to avoid checking the transfer value in
gst_v4l2_video_colorimetry_matches when it is
GST_VIDEO_TRANSFER_ARIB_STD_B67. This is because the transfer value in
the cinfo parameter comes from gst_v4l2_object_get_colorspace, which
converts the transfer to another value, causing a mismatch.
Since the kernel does not support GST_VIDEO_TRANSFER_ARIB_STD_B67,
gst_v4l2_object_get_colorspace cannot map it correctly from V4L2 to
GStreamer. Therefore, we ignore this check to prevent errors.
changes:
- Added a condition in gst_v4l2_video_colorimetry_matches to bypass the
transfer check when the transfer is GST_VIDEO_TRANSFER_ARIB_STD_B67.
- Ensured that the pipeline does not throw errors due to unsupported
bt2100-hlg colorimetry in V4L2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7212>
With GLES 2.0 we are forced to use CopyTextImage2D which requires
passing an internal format. With QT6 eglfs, we need to pass GL_RGB
instead, probably because of how the texture has been created. As its
hard to guess, simply fallback to GL_RGB on failure. This fixes usage
or qml6glsrc with eglfs backend, without loosing support for
semi-transparent window on other platforms.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7321>
