- generate a new session id for every new client
use the session id in the resource url
- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer
- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id
- Add a new structure Session
- manage each producer using a session
- avoid send EOS when a session terminates, instead keep running
waiting for any new producer to connect
- Maintain a bin element per session
- each session bin encapsulates webrtcbin and the decoder if needed
as well as the parser and filter if requested by the application
(through request-encoded-filter)
- this will be helpful to cleanup the session's respective elements
when the corresponding producer terminates the session
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>
Clippy caught the missing feature `signal` which is used by the WebRTC precise
synchronization examples. When running `cargo` `check`, `build` or `clippy`
without `no-default-dependencies`, this feature was already present due to
dependents crates.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1541>
When swapping between several development branches, compilation times can be
frustrating. This commit proposes adding features to control which signaller
to include when building the webrtc plugin. By default, all signallers are
included, just like before.
Compiling the `webrtc-precise-sync` examples with `--no-default-features`
reduces compilation to 267 crates instead of 429 when all signallers are
compiled in.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1539>
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization)
for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to
identify the clock. These attributes are handled by `rtpjitterbuffer` on the
consumer side. They MUST be part of the SDP offer.
When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the
clock's origin. Because the payloaders are not instantiated when the offer is
sent to the consumer, the RTP offset is set to 0 and the payloader
`timstamp-offset`s are set accordingly when they are created.
The `webrtc-precise-sync` examples were updated to be able to start with an NTP
(default), a PTP or the system clock (on the receiver only). The rtp jitter
buffer will synchronize with the clock signalled in the SDP offer provided the
sender is started with `--do-clock-signalling` & the receiver with
`--expect-clock-signalling`.
[RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
When an encoder was not supported by the `VideoEncoder` `bitrate` accessors, an
`unimplemented` panic would occur which would poison `state` & `settings`
`Mutex`s resulting in other threads panicking, notably entering `end_session()`,
which lead to many failures in `BinImplExt::parent_remove_element()` until a
segmentation fault ended the process. This was observed using `vaapivp9enc`.
This commit logs a warning if an encoder isn't supported by the `bitrate`
accessors and silently by-passes `bitrate`-related operations when unsupported.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1534>
Otherwise the clock id will simply be overridden instead of unscheduling
it, and if the streaming thread of the source pad currently waits on it
then it will wait potentially for a very long time and deactivating the
pad would wait for that to happen.
Also unschedule the clock id on `Drop` of the state to be one the safe
side and not simply forget about it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1526>
Add `self.increasing_duration` and `self.increasing_counter`
to logs to provide more details of why `overuse_filter()`
determines overuse of network.
To get access to the latest values of those fields we need
to move down the log call. But that is fine, since no other
logged data is modified between the old and new location of
`gst::log!()`.
We do not bother logging `self.last_overuse_estimate` since
that is simply the previously logged value of `estimate`. We
must put the log call before we write the latest value to it
though, in case we want to log it in the future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1522>
Quoting [`BehaviorVersion` documentation]:
> Over time, new best-practice behaviors are introduced. However, these
> behaviors might not be backwards compatible. For example, a change which
> introduces new default timeouts or a new retry-mode for all operations might
> be the ideal behavior but could break existing applications.
This commit uses `BehaviorVersion::v2023_11_09()`, which is the latest
major version at the moment. When a new major version is released, the method
will be deprecated, which will warn us of the new version and let us decide
when to upgrade, after any changes if required. This is safer that using
`latest()` which would silently use a different major version, possibly
breaking existing code.
[`BehaviorVersion` documentation]: https://docs.rs/aws-config/1.1.8/aws_config/struct.BehaviorVersion.html
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1520>
The following error is logged when `webrtcsink` is feeded with an audio stream:
> ERROR video-info video-info.c:540:gst_video_info_from_caps:
> wrong name 'audio/x-raw', expected video/ or image/
This commit bypasses `VideoInfo::from_caps` for audio streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1511>
Some elements in the RTP stack assume all buffers in a `gst::BufferList`
correspond to the same timestamp. See in [`rtpsession`] for instance.
This also had the effect that `rtpsession` did not create correct RTCP as it
only saw some of the SSRCs in the stream.
`rtpgccbwe` formed a packet group by gathering buffers in a `gst::BufferList`,
regardless of whether they corresponded to the same timestamp, which broke
synchronization under certain circonstances.
This commit makes `rtpgccbwe` push the buffers as they were received: one by one.
[`rtpsession`]: bc858976db/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c (L2462)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1502>
Instead of exposing all ids properties as strings, we now have two
signaller implementations exposing those properties using their actual
type. This API is more natural and save the element and application
conversions when using numerical ids (Janus's default).
I also removed the 'joined-id' property as it's actually the same id as
'feed-id'. I think it would be better to have a 'janus-state' property or
something like that for applications willing to know when the room has
been joined.
This id is also no longer generated by the element by default, as Janus
will take care of generating one if not provided.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1486>
Without sending a Leave request to the server before disconnecting, the
disconnected client will appear present and stuck in the room for a little
while until the server removes it due to inactivity.
After this change, the disconnecting client will immediately leave the room.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1482>
