- From: Sergio Garcia Murillo via GitHub <sysbot+gh@w3.org>
- Date: Mon, 20 Jul 2020 21:08:38 +0000
- To: public-webrtc-logs@w3.org
My two cents. I am currently working on an use case that requires synchronized playback of a remote stream in two different devices. We add a custom delay on the primary node via web audio and then adjust the secondary via `playoutDelayHint` and adjust this based on the `rtt`. So, having a numeric value makes sense, at least for us. Regarding the time required to adjust the delay to the new value, it is an implementation detail of NetEq. We have modified it so it converges faster (whiting 1 second) to the value set by js by adding silence or dropping packets instead of the default behavior of NetEq. On a side note, I would be awesome if we could add more parameters to control the jitter buffer behavior (or even completely replace it) as, at least NetEq, is not tuned correctly for several use cases. -- GitHub Notification of comment by murillo128 Please view or discuss this issue at https://github.com/w3c/webrtc-extensions/issues/46#issuecomment-661334298 using your GitHub account
Received on Monday, 20 July 2020 21:08:40 UTC