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Re: [webrtc-stats] Stats to keep track of sync between audio and video

From: Stefan Holmer via GitHub <sysbot+gh@w3.org>
Date: Tue, 07 Feb 2017 14:56:49 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-278023581-1486479407-sysbot+gh@w3.org>
googCaptureStartNtpTimeMs delay is based on an estimate of the time 
the capture started on the clock of the receiving endpoint, right? If 
remoteVideo.currentTime would represent the estimated current time of 
the capturer in the receivering endpoints (local) clock, then I think 
that might be a good idea.

I'd like the playout timestamp to be currentPlayoutTimestamp, which 
represents the timestamp which should have been played out at this 
very moment if there was something to play, and not the timestamp 
which was last played out. Otherwise we'll have to correct for the 
time that has passed since the last playout, which will complicate 
things (and we may need yet another timestamp for that).

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Received on Tuesday, 7 February 2017 14:56:56 UTC

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