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 the UVC gadget announces multiple formats in the descriptors the uvcsink
doesn't select the actual format but let's the UVC hosts select the format.
If the GStreamer pipeline is started before a UVC host selected the format,
upstream decides on a format until the UVC host has decided. In this case, the
current format needs to be set based on the caps from the caps event to be able
to detect if the format selection by the UVC host requires a format change on
the GStreamer pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7473>
The uvcsink may be put into the READY state to start listening for UVC requests.
Therefore, the UVC host may set a streaming format before the GStreamer pipeline
is started and the uvcsink received a caps event. In this case, prev_caps will
be NULL.
If the EVENT_CAPS has not been received, skip the check if the format needs to
be changed, since the sink will be started with the format selected by the UVC
host, anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7473>
Until now we were overriding pad functions forgetting about the function
data (that are set using the _full variant of the functions setters), meaning
that the data was lost and any user of that feature would get empty data when
the wrapped function were called.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7466>
Adding a property to control the number of in-flight GPU commands
(default is unlimited). Note that actual maximum number is defined
in d3d12device's direct command queue object which is 32 now,
thus total number of scheduled GPU commands cannot exceed 32.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7444>
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>
Sometimes under certain loads, VT can error out with kVTVideoEncoderMalfunctionErr or kVTVideoEncoderNotAvailableNowErr.
These have been reported to happen more often than usual if CopyProperty/SetProperty() is used close to the encode call.
Both can be worked around by restarting the encoding session.
These errors can be returned either directly from VTCompressionSessionEncodeFrame() or later in the encoding callback.
This patch handles both scenarios the same way - a session restart is be attempted on the next encode_frame() call.
If the error is returned immediately by the encode call, it's possible that some correct frames will still be given to
the output callback, but for simplicity (+ because I wasn't able to verify this scenario) let's just discard those.
In addition, this commit also simplifies the beach/drop logic in enqueue_buffer.
Related bug reports in other projects:
http://www.openradar.me/45889262https://github.com/aws/amazon-chime-sdk-ios/issues/170#issuecomment-741908622
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7173>
If state is changing from playing to paused, and rate is reset to 1
which causes seek position is valid, current code will do seek for
streams that are not seekable. So need to check whether stream is
seekable before seeking.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7441>
All accesses to it were protected either by a mutex already, or at least
used yet another mutex for gst_poll_read_control() / gst_poll_write_control().
The usage of GstPoll has to stay for backwards compatibility as it is
used to manage the (public) fd that can be used to wait for the bus to
be ready, but this switch at least simplifies the implementation a bit
and results in fewer atomic operations.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6684>
It was iterating over each field and after fixating its value was again
iterating over every field to find where to store the value.
Instead directly overwrite the value after validating it.
Also actually check that the structure is writable before modifying its fields
by using gst_structure_map_in_place() instead of gst_structure_fixate().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7420>
If the pending remote description has an invalid BUNDLE group _parse_bundle()
triggers early return from _create_answer_task(), before ret has been
initialized, so it needs to be checked before attempting to call
gst_sdp_message_copy().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7423>
The initial calculation for the precision shift was wrong and would allow for
overflows during the calculations which were not detected and lead to wrong
results.
Also add a test for a case where overflows where previously not detected and
caused a completely wrong result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7406>
webrtcsrc first creates recvonly transceivers with codec-preferences
and expects that after applying a remote description, the
previously created transceivers are used rather than having new
transceivers created.
When pairing webrtcsink + webrtcsrc, the offer sdp from webrtcsink has a media
section with sendonly direction. In !7156, which was implemented following
RFC9429 Section 5.10, we only reuse a unassociated transceiver when applying a
remote description if the media is sendrecv or recvonly, and that caused creation
of new transceivers when applying a remote offer in webrtcsrc, thus losing
information from codec preferences like the RTP extension headers in the
previously created transceivers.
Since the change in !7156 broke existing code from webrtcsrc, relax the condition
for reusing unassociated transceivers and add a test to document this behavior which
wasn't covered by any tests before.
Fixes#3753.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7417>
```
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>
Before trying to retrieve a GMainContext from a provided
GstPlayerSignalDispatcher, check that it is actually
GstPlayerGMainContextSignalDispatcher. If not, use the
default GMainContext for dispatching signals via the adapter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7392>
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>
Don't reuse the same stats state structure across multiple
get-stats calls. Make each callback take a copy of the
non-changing fields it needs and use a local working copy
to avoid crashing.
Fixes problems with the unit test crashing sometimes for the
unit test introduced in MR !7338
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7387>
Since bins can set the context of their children elements, the set_context()
vmethod shouldn't call bus messages post methods, since it locks the parent
object, the bin, which might be already locked, leading to a deadlock.
Fixes: #3706
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7378>
When multiple streams are bundled on the same transport,
the statistics would end up incorrectly generated,
as each pad would regenerate stats for every ssrc on the
transport, overwriting previous iterations and assigning
bogus media kind and other values to the wrong ssrc.
Fix by making sure each pad only loops and generates
statistics for the one ssrc that pad is receiving / sending.
Add a unit test that the codec kind field in RTP statistics
are now generated correctly.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2555
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7338>
Adding a new videosink element for Windows composition API based
applications. Unlike d3d12videosink, this element will create only
DXGI swapchain by using IDXGIFactory2::CreateSwapChainForComposition()
without actual window handle, so that video scene can be composed
via Windows native composition API, such as DirectComposition.
Note that this videosink does not support GstVideoOverlay interface
because of the design.
The swapchain created by this element can be used with
* DirectComposition's IDCompositionVisual in Win32 app
* WinRT and WinUI3's UI.Composition in Win32/UWP app
* UWP and WinUI3 XAML's SwapChainPanel
See also examples in this commit which show usage of the videosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7287>
Prevent the default webrtc test machinery from attempting to
create and set an answer when we're just testing rollback
of the offers. Add some locking / waiting to ensure the test
is complete before exiting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7365>
