They're not recommended by the spec to include in the RTP packets but it
is valid to include them. Pion is including them.
When parsing the size fields also make sure to only take that much of a
payload unit and to skip any trailing data (which should not exist in
the first place).
Pion is also currently storing multiple OBUs in a single payload unit,
which is not allowed by the spec but can be easily handled with this
code now.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/560
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1621>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if cancellation should happen and reset this
later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1617>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Also implement actual unlocking in s3hlssink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1617>
It would be possible that there is no cancellable yet when unlock() is
called, then the setup task is started and it would simply run and being
waited on instead of not being run at all.
To solve this, remember if unlock() was called and reset this in
unlock_stop().
Also make sure to not keep the abort handle locked while waiting,
otherwise cancellation would never actually work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1617>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1617>
Otherwise when the segment durations aren't as clean cut as in the
example, multiple segments with the exact same duration in milliseconds
will get output, even though they could have been repeated.
Fix this so that people copying this code don't encounter the bug.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
While monitoring and debugging rtpgccbwe, it is very helpful
to get continuous values of what it considers the effective
bitrate. Right now such prints will stop coming once the
algorithm stabilizes. Print it in more places so it keeps
coming. Use the same format to make it simpler to extract
the values by parsing the logs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
When debugging rtpgccbwe it is helpful to know if it is
delay based or loss based band-width estimation that puts a
bound on the current target bitrate, so add logs for that.
To minimize the time we need to hold the state lock, perform
the logging after we have released the state lock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
If force-aspect-ratio=false then make sure to fully fill the given
width/height with the video frame and avoid rounding errors. This makes
sure that the video is rendered in the exact position selected by the
caller and that graphics offloading is going to work more likely.
In other cases and for all overlays, make sure that the calculated
positions are staying inside (0, 0, width, height) as rendering outside
is not allowed by GTK.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1556>
If a user constrained the supported CAPS, for instance using `video-caps`:
```shell
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420 ! x264enc \
! webrtcsink video-caps=video/x-vp8
```
... a panic would occur which was internally caught without the user being
informed except for the following message which was written to stderr:
> thread 'tokio-runtime-worker' panicked at net/webrtc/src/webrtcsink/imp.rs:3533:22:
> expected audio or video raw caps: video/x-h264, [...] <br>
> note: run with `RUST_BACKTRACE=1` environment variable to display a backtrace
The pipeline kept running.
This commit converts the panic into an `Error` which bubbles up as an element
`StreamError::CodecNotFound` which can be handled by the application.
With the above `gst-launch`, this terminates the pipeline with:
> [...] ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:3771:gstrswebrtc::
> webrtcsink:👿:BaseWebRTCSink::start_stream_discovery_if_needed::{{closure}}:<webrtcsink0>
> Error running discovery: Unsupported caps: video/x-h264, [...] <br>
> ERROR: from element /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
> There is no codec present that can handle the stream's type. <br>
> Additional debug info: <br>
> net/webrtc/src/webrtcsink/imp.rs(3772): gstrswebrtc::webrtcsink:👿:BaseWebRTCSink::
> start_stream_discovery_if_needed::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
> Failed to look up output caps: Unsupported caps: video/x-h264, [...] <br>
> Execution ended after 0:00:00.055716661 <br>
> Setting pipeline to NULL ...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1556>
