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 accountReceived on Thursday, 14 February 2019 14:43:33 UTC
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