Contact emails
zst...@chromium.org, hb...@chromium.org
Spec
https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver
https://w3c.github.io/webrtc-pc/#dom-rtcrtpsynchronizationsource
Summary
This intent is to implement and ship more of the receiving part of the RTP Media API (old intent to implement and ship), specifically RTCRtpSynchronizationSource. The implemented portion will be the same as currently exists for RTCRtpContributingSource (intent to implement and ship).
RTCRtpSender extension:
- sequence<RTCRtpSynchronizationSource> getSynchronizationSources();
RTCRtpSynchronizationSource support:
- DOMHighRestTimeStamp timestamp;
- unsigned long source;
- byte audioLevel;
Motivation
The W3C WebRTC working group has consensus around adding this API.
Will this feature be supported on all six Blink platforms (Windows, Mac, Linux, Chrome OS, Android, and Android WebView)?
Yes.
Is this feature fully tested by web-platform-tests?
The tests are already upstream: https://github.com/w3c/web-platform-tests/blob/master/webrtc/RTCRtpReceiver-getSynchronizationSources.https.html
Link to entry on the feature dashboard
Previous intents link to these features:
RTCRtpSender and RTCRtpReceiver extensions to webkitRTCPeerConnection (https://www.chromestatus.com/feature/5347809238712320)
Partial RTCRtpReceiver and RTCRtpContributingSource support (https://www.chromestatus.com/feature/5715393821802496)
I could either extend the scope of one of those, or add a new feature specific to RTCRtpSynchronizationSource. Please advise.
Requesting approval to ship?
Yes.
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Status entry: https://www.chromestatus.com/feature/5637113429098496
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an audio mixer or forwarder receives audio streams from many or all of the conference participants. It then selectively forwards some of them to other participants in the conference. In large conferences, it is possible that such a server might be receiving a large number of streams, of which only a few are intended to be forwarded to the other conference participants.
RTCRtpContributingSource.audioLevel assumes a different architecture as described in RFC 6465:
"[The mixer is] responsible for combining the media streams that make up a conference, and generating one or more output streams that are delivered to recipients". Every participant would hence receive, in a flat single stream, media originating from all the others.[...] The flat nature of the streams that amixer would output and send to participants makes it difficult for users to identify the original source of what they are hearing.Here, the audioLevel could be used by a receiving client to infer e.g. the current speaker.So I think it is reasonable to implement them separately. RTCRtpContributingSource.audioLevel will also require some additional work in WebRTC before it can be implemented here.Audio level is optional for contributing sources because it isn't computable by the browser if it isn't included in the header extension (the browser only gets one flat stream, so it can't tell how much of the audio came from various senders).Synchronization sources don't have this problem, so the audio level is always computable. That being said, the initial implementation will make the field nullable and only include it if the header extension is present (like you suggest).
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