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

From: Taylor Brandstetter via GitHub <sysbot+gh@w3.org>
Date: Thu, 16 Mar 2017 22:23:34 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-287209355-1489703013-sysbot+gh@w3.org> 
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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Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1085#issuecomment-287209355 using your GitHub account
Received on Thursday, 16 March 2017 22:23:40 UTC

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