[webrtc-pc] Add RTCRtpSender.getSynchronizationSources() to expose audioLevel (#2103)

henbos has just created a new issue for https://github.com/w3c/webrtc-pc:

== Add RTCRtpSender.getSynchronizationSources() to expose audioLevel ==
A lot of people poll getStats() frequently to obtain audio levels, which is supposedly inefficient to the point that it can make your laptop sound like a helicopter. We already have RTCRtpReceiver.getSynchronizationSources()[i].audioLevel, should we add getSynchronizationSources() to RTCRtpSender as well? It would represent the RTP packets sent instead of the ones received, so you can read the local audio levels without setting up a loopback connection.

Related question:
- Is this audioLevel "good enough" to replace getStats()'s audio level? This is an instantaneous value, and while it can be polled frequently without any significant overhead, it may be based on something other than what people are used to.

Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2103 using your GitHub account

Received on Thursday, 14 February 2019 13:35:06 UTC