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Re: [webrtc-pc] RTCRtpContributingSource.audioLevel not guaranteed to be in sync with audio playout

From: Taylor Brandstetter via GitHub <sysbot+gh@w3.org>
Date: Thu, 23 Mar 2017 01:28:29 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issue_comment.created-288590235-1490232508-sysbot+gh@w3.org>
Another issue that was brought to my attention recently: this part of the description of `audioLevel` means that implementations are required to decode a packet and compute the audio level as soon as a packet is received:

> If an RFC 6464 extension header is not present, the browser will compute the value as if it had come from RFC 6464 and use that.

Doing this would be bad for performance. Chrome currently only decodes a packet and computes the audio level (for `getStats`) when more data is needed for playout.

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Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/1085#issuecomment-288590235 using your GitHub account
Received on Thursday, 23 March 2017 01:28:35 UTC

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