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>
