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Author SHA1 Message Date
Sebastian Dröge dd4c466265 deny: Add override for older tungstenite 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1616>
 2024-06-14 05:46:18 +00:00
Sebastian Dröge 1a03edd27d webrtc: Update to async-tungstenite 0.26 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1616>
 2024-06-14 05:46:18 +00:00
Sebastian Dröge 61d2259b6b webrtchttp: Fix race condition when unlocking 
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>
 2024-06-13 13:19:41 +00:00
Sebastian Dröge d7b2f15df6 aws: Fix race condition when unlocking 
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>
 2024-06-13 13:19:41 +00:00
Sebastian Dröge 6fd184cc96 reqwesthttpsrc: Fix race condition when unlocking 
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>
 2024-06-13 13:19:41 +00:00
Sebastian Dröge ec26fcc65a ndi: Add support for loading NDI SDK v6 
The library name and environment variable name have changed but the ABI
is completely compatible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1615>
 2024-06-13 13:01:11 +00:00
Sebastian Dröge ac79b52cff rtp: Don't restrict payload types for payloaders 
WebRTC uses payload types 35-63 as dynamic payload types too to be able
to place more codec variants into the SDP offer.

Instead of allowing just certain payload types, completely remove any
restrictions and let the user decide. There's technically nothing wrong
with using any payload type, especially when using the encoding-name.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1595>
 2024-06-13 12:37:52 +00:00
Francisco Javier Velázquez-García 205cc10e6e webrtcsink: Refactor value retrieval to avoid lock poisoning 
When setting an incorrect property name in settings,
start_stream_discovery_if_needed would panic because it attempts to
unwrap a poisoned lock for settings.

This refactor avoids that situation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1598>
 2024-05-31 12:41:45 +03:00
Francisco Javier Velázquez-García 3cdb350626 webrtcsink: Fix typo in property name for av1enc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1598>
 2024-05-31 12:41:45 +03:00
cdelguercio 88b8e35871 webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1597>
 2024-05-31 11:18:08 +03:00
cdelguercio 32b987d73e webrtcsink: Add VP9 parser after the encoder for VP9 too 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:21:54 +03:00
Sebastian Dröge 73158bf58b aws: Update to base32 0.5 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:21:25 +03:00
Robert Ayrapetyan 3036cadd20 webrtc: add support for insecure tls connections 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:19:26 +03:00
Martin Nordholts d0dc85293d rtpgccbwe: Also log self.measure in overuse_filter() 
Also log `self.measure` in overuse_filter() since tracking
`self.measure` over time help a lot in making sense of
`self.estimate` (and `amplified_estimate`).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:18:01 +03:00
Martin Nordholts be72988545 rtpgccbwe: Rename variable t to amplified_estimate 
We normally multiply `self.estimate` with `MAX_DELTAS` (60).
Rename the variables that holds the result of this
calculation to `amplified_estimate` to make the distinction
clearer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:17:56 +03:00
Martin Nordholts 536601e65d rtpgccbwe: Log effective bitrate in more places 
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>
 2024-05-23 16:14:20 +03:00
Martin Nordholts 26f63c5e1e rtpgccbwe: Add mising 'ps' suffix to 'kbps' of 'effective bitrate' 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:14:13 +03:00
Martin Nordholts 3a6c663f3c rtpgccbwe: Log delay and loss target bitrates separately 
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>
 2024-05-23 16:13:53 +03:00
Mathieu Duponchelle e72e361b63 webrtcsink: improve error when no discovery pipeline runs 
If for instance no encoder was found or the RTP plugin was missing,
it is possible that no discovery pipeline will run for a given stream.

Provide a more helpful error message for that case.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/534
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:13:42 +03:00
Sebastian Dröge 1bee96ccb4 Fix new Rust 1.78 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:13:15 +03:00
Sebastian Dröge b498c44df5 rtpgccbwe: Move away from deprecated time::Instant to std::time::Instant 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1556>
 2024-04-29 11:50:29 +03:00
François Laignel e1b8b8befd webrtcsink: don't panic if input CAPS are not supported 
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>
 2024-04-29 11:50:12 +03:00
François Laignel 47429e2ed8 gccbwe: don't log an error when handling a buffer list while stopping 
When `webrtcsink` was stopped, `gccbwe` could log an error if it was handling a
buffer list. This commit logs an error only if `push_list()` returned an error
other than `Flushing`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1556>
 2024-04-29 11:49:21 +03:00
Sanchayan Maity 6d36b9160a aws: Introduce a property to use path-style addressing 
AWS SDK switched to virtual addressing as default instead of path
style earlier. While MinIO supports virtual host style requests,
path style requests are the default.

Introduce a property to allow the use of path style addressing if
required.

For more information, see
https://github.com/minio/minio/blob/master/docs/config/README.md#domain
https://docs.aws.amazon.com/AmazonS3/latest/userguide/VirtualHosting.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1556>
 2024-04-29 11:49:04 +03:00
François Laignel 0c55ac9e31 webrtcsink: don't panic with bitrate handling unsupported encoders 
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/1556>
 2024-04-29 11:48:41 +03:00
Taruntej Kanakamalla 20380e699e webrtcsrc: change the producer-id type for request-encoded-filter 
With https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1477
the producer id used while emitting the request-encoded-filter
can be a None if the msid of the webrtcbin's pad is None.
This might not affect the signal handler written in C but
can panic in an existing Rust application with signal
handler which can only handle valid String type as its param
for the producer id.

So change the param type to Option<String> in the signal builder
for request-encoded-fiter signal

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1556>
 2024-04-29 11:48:25 +03:00
Sebastian Dröge 58ae74467a webrtchttp: Update to reqwest 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:19:45 +03:00
Sebastian Dröge 0b0fc7f7d4 onvifmetadataparse: Reset state in PAUSED->READY after pad deactivation 
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/1531>
 2024-04-08 15:19:21 +03:00
Taruntej Kanakamalla 98dce3c49c net/webrtc: fix inconsistencies in documentation of object names 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:19:08 +03:00
Guillaume Desmottes 3c71247ac9 web: webrtcsink: improve panic message on unexpected caps during discovery 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:18:51 +03:00
Guillaume Desmottes 762d4a4437 webrtc: webrtcsink: set perfect-timestamp=true on audio encoders 
Chrome audio decoder doesn't cope well with not perfect ts, generating
noises in the audio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:18:43 +03:00
Martin Nordholts 0faa74b74d rtpgccbwe: Add increasing_duration and counter to existing gst::log!() 
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/1531>
 2024-04-08 15:18:03 +03:00
François Laignel d0461b190f aws: improve error message logs 
The `Display` and `Debug` trait for the AWS error messages are not very useful.

- `Display` only prints the high level error, e.g.: "service error".
- `Debug` prints all the fields in the error stack, resulting in hard to read
  messages with redudant or unnecessary information. E.g.:

> ServiceError(ServiceError { source: BadRequestException(BadRequestException {
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), meta: ErrorMetadata { code: Some("BadRequestException"),
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), extras: Some({"aws_request_id": "1b8bbafd-5b71-4ba5-8676-28432381e6a9"}) } }),
> raw: Response { status: StatusCode(400), headers: Headers { headers:
> {"x-amzn-requestid": HeaderValue { _private: H0("1b8bbafd-5b71-4ba5-8676-28432381e6a9") },
> "x-amzn-errortype": HeaderValue { _private:
> H0("BadRequestException:http://internal.amazon.com/coral/com.amazonaws.transcribe.streaming/") },
> "date": HeaderValue { _private: H0("Tue, 26 Mar 2024 17:41:31 GMT") },
> "content-type": HeaderValue { _private: H0("application/x-amz-json-1.1") },
> "content-length": HeaderValue { _private: H0("315") }} }, body: SdkBody {
> inner: Once(Some(b"{\"Message\":\"1 validation error detected: Value 'test'
> at 'languageCode' failed to satisfy constraint: Member must satisfy enum value
> set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT,
> es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP,
> ca-ES, es-US, fr-CA, en-GB]\"}")), retryable: true }, extensions: Extensions {
> extensions_02x: Extensions, extensions_1x: Extensions } } })

This commit adopts the most informative and concise solution I could come up
with to log AWS errors. With the above error case, this results in:

> service error: Error { code: "BadRequestException", message: "1 validation
> error detected: Value 'test' at 'languageCode' failed to satisfy constraint:
> Member must satisfy enum value set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI,
> pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN,
> en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US, fr-CA, en-GB]",
> aws_request_id: "a40a32a8-7b0b-4228-a348-f8502087a9f0" }

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:17:56 +03:00
François Laignel 85ce07d184 aws: use fixed BehaviorVersion 
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/1531>
 2024-04-08 15:17:49 +03:00
Sebastian Dröge b05436d6d3 reqwest: Update to reqwest 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:17:31 +03:00
Philippe Normand 23403b5c9a Fix clippy warnings after upgrade to Rust 1.77 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:15:26 +03:00
François Laignel fd154d272c webrtcsrc: add do-retransmission property 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
François Laignel ede354d7a5 webrtcsink: prevent video-info error log for audio streams 
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/1508>
 2024-03-21 13:30:20 +02:00
François Laignel 158fe80779 rtp: gccbwe: don't break downstream assumptions pushing buffer lists 
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/1508>
 2024-03-21 13:30:20 +02:00
Guillaume Desmottes a502dba6d5 webrtc: janus: handle 'hangup' messages from Janus 
Fix error about this message not being handled:

{
   "janus": "hangup",
   "session_id": 4758817463851315,
   "sender": 4126342934227009,
   "reason": "Close PC"
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
Guillaume Desmottes be055f6dfa webrtc: janus: handle 'destroyed' messages from Janus 
Fix this error when the room is destroyed:

ERROR   webrtc-janusvr-signaller imp.rs:413:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x55b166a3fe40> Unknown message from server: {
   "janus": "event",
   "session_id": 6667171862739941,
   "sender": 1964690595468240,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "destroyed",
         "room": 8320333573294267
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
Sebastian Dröge c982db73a7 rtp: Switch from chrono to time 
Which allows to simplify quite a bit of code and avoids us having to
handle some API deprecations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
Guillaume Desmottes 21f59c65da webrtc: allow resolution and framerate input changes 
Some changes do not require a WebRTC renegotiation so we can allow
those.

Fix #515

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz 7a90e96332 livekit_signaller: Added missing getter for excluded-producer-peer-ids 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz 283d1568b4 webrtcsrc: Removed incorrect URIHandler from LiveKit source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz 51287705ce livekit_signaller: Improved shutdown behavior 
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/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz e9edee131b webrtcsrc: Removed flag setup from WhipServerSrc 
It's already done in the base class

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz feb01510f9 webrtcsrc: Updated readme for LiveKit source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz 32e13f0a10 webrtcsrc: Added LiveKit source element 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz c8dcd50846 webrtcsink: Updated livekitwebrtcsink for new signaller constructor 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00