Only configure header extensions from the source pad caps if they exist
already in the source pad caps, otherwise the configuration will fail.
Extensions that are added via the signals might not exist in the source
pad caps yet and would be added later.
Also, if configuring an existing extension from the new caps fails,
remove it and try to request a new extension for it.
Additionally don't remove extensions from the caps that can't be
provided. No header extensions for them would be added to the packets,
but that's not a problem. Removing them on the other hand would cause
negotiation to fail. This only affects extensions that are already
included in the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
If configuring an existing extension from the new caps fails, remove it
and try to request a new extension for it.
Also remove all extensions from the list that are not provided in the
caps, instead of passing RTP packets to all of them anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
In our tests, the slope (found with linear regression) on a
history of the (smoothed) accumulated inter-group delays
gives a more stable congestion control. In particular,
low-end devices becomes less sensitive to spikes in
inter-group delay measurements.
This flavour of delay based bandwidth estimation with Google
Congestion Control is also what Chromium is using.
To make it easy to experiment with the new estimator, as
well as add support for new ones in the future, also add
infrastructure for making delay estimator flavour selectable
at runtime.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
Rust targets without support for `AtomicU64` is still
somewhat common. Running
git grep -i 'max_atomic_width: Some(32)' | wc -l
in the Rust compiler repo currently counts to 34 targets.
Change the `RtpBasePay2::ssrc_collision` from `AtomicU64` to
`Mutex<Option<u32>>`. This way we keep support for these
targets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1562>
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/1567>
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/1561>
This set of changes implements the below fixes:
- Allow certificates to be specified for client/quicsink
- Secure connection being true on server/quicsrc and false on
client/quicsink still resulted in a successful connection
instead of server rejecting the connection
- Using secure connection with ALPN was not working
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
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/1540>
