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Re: [webrtc-pc] Attempt to update RTCRtpContributingSource objects at playout time.

From: Taylor Brandstetter via GitHub <sysbot+gh@w3.org>
Date: Wed, 05 Apr 2017 00:45:18 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-291692694-1491353117-sysbot+gh@w3.org>
I understand the concern. Maybe calling it "playout time" is the mistake, but what we ultimately need is _some_ well-defined, post-jitter-buffer point of time, which the application has a way of correlating to playout time.

Even if the application does something with the MediaStreamTrack that introduces extra delay, the original problem is solved as long as the application has a way of estimating the additional delay and factoring it in when updating its audio level UI.

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GitHub Notification of comment by taylor-b
Please view or discuss this issue at https://github.com/w3c/webrtc-pc/pull/1098#issuecomment-291692694 using your GitHub account
Received on Wednesday, 5 April 2017 00:45:25 UTC

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