Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout

Another option: add a method on `RTCRtpReceiver` to get the remote timestamp of the last frame that was played out.

An application could call `getContributingSources` and see a source with `timestamp` X, call `getCurrentPlayoutTimestamp` and get Y, and then wait for Y - X before updating the audio level UI.

The advantages of this approach are that it's simpler from an implementation perspective, and it allows the application to get information sooner, in case that's ever desired.

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Received on Thursday, 16 March 2017 22:23:40 UTC