- From: henbos via GitHub <sysbot+gh@w3.org>
- Date: Tue, 14 Feb 2017 15:40:38 +0000
- To: public-webrtc-logs@w3.org
+@hlundin Is this right? + **Jitter buffer delay**: The currently expected delay due to jitter buffer, which is the difference between the timestamp of the latest audio sample coming out of the jitter buffer and the timestamp it had when it was inserted into the jitter buffer. + **Sound card delay**: This has to do with the size of the buffer we feed to the sound card, since the entire buffer has to be filled up before fed to the sound card. The delay is (_number of samples that fit into the buffer_ - 1) * (_the time it takes to play one sample_). + **Algorithmic delay**: _???_ If we get these right we can define `RTCMediaStreamTrackStats.audioDelay` as the sum of these delays. -- GitHub Notification of comment by henbos Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/151#issuecomment-279742750 using your GitHub account
Received on Tuesday, 14 February 2017 15:40:44 UTC