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

From: Harald Alvestrand via GitHub <sysbot+gh@w3.org>
Date: Thu, 14 Feb 2019 14:43:32 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-463652202-1550155411-sysbot+gh@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.

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

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