- From: Elad Alon via GitHub <sysbot+gh@w3.org>
- Date: Thu, 26 Sep 2019 10:30:56 +0000
- To: public-webrtc-logs@w3.org
eladalon1983 has just created a new issue for https://github.com/w3c/webrtc-pc: == Add adaptivePtime to RTCConfiguration == Currently, there is no way to tell the browser that it should attempt to optimize the trade-off between audio latency and bandwidth consumption, by varying the length of audio frames dynamically during a call. It would be good to add this. See [this document](https://docs.google.com/document/d/12sVXRog-Dl9c-hbMInz9obmODk_yft06z7GMgfpdjPw/edit?usp=sharing) for details. Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2310 using your GitHub account
Received on Thursday, 26 September 2019 10:30:58 UTC