- From: Taylor Brandstetter via GitHub <sysbot+gh@w3.org>
- Date: Wed, 05 Apr 2017 00:45:18 +0000
- To: public-webrtc-logs@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. -- 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