- From: Harald Alvestrand via GitHub <sysbot+gh@w3.org>
- Date: Thu, 14 Feb 2019 14:43:32 +0000
- To: public-webrtc-logs@w3.org
This need should be solved by reading back audio level on a MediaStreamTrack instead. It's more general, and doesn't depend on the vagaries of the particular RTP packetization regime in force. If we ever get to sending pre-encoded audio streams, supporting sender's audio level would require re-decoding the audio packets in order to read the level; that sounds like an architecturally wrong approach. -- GitHub Notification of comment by alvestrand Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2103#issuecomment-463652202 using your GitHub account
Received on Thursday, 14 February 2019 14:43:33 UTC