Re: [webrtc-stats] Stat for audio playout delay


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.

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Received on Tuesday, 14 February 2017 15:40:44 UTC