Background: RTCRtpReceiver's[1] getSynchronizationSources() method
allows WebRTC applications to monitor volume levels of call participants
without having to use the costly getStats()[2] call. RTCRtpReceiver's
getContributingSources() method enables WebRTC applications to monitor
the volume level of participants whose audio is being mixed upstream if
the mixer includes this data in the RTP extension headers.
We intend to ship these interfaces in 59, and they are on nightly now.
Tracking: https://bugzilla.mozilla.org/show_bug.cgi?id=1363667
[1] https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver
[2] https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats()
Cheers,
Nico Grunbaum
-. --.
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