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[webrtc-stats] RTCMediaStreamTrackStats.audioLevel... input vs output?

From: henbos via GitHub <sysbot+gh@w3.org>
Date: Tue, 31 Jan 2017 16:05:06 +0000
To: public-webrtc@w3.org
Message-ID: <issues.opened-204341841-1485878703-sysbot+gh@w3.org>
henbos has just created a new issue for 
https://github.com/w3c/webrtc-stats:

== RTCMediaStreamTrackStats.audioLevel... input vs output? ==
The spec currently says:
> Only valid for audio. The value is between 0..1 (linear), where 1.0 
represents 0 dBov, 0 represents silence, and 0.5 represents 
approximately 6 dBSPL change in the sound pressure level from 0 dBov.
> 
> The "audio level" value defined in [RFC6464] and used in the 
RTCRtpContributingSource.audioLevel of [WEBRTC] (defined as 0..127, 
where 0 represents 0 dBov, 126 represents -126 dBov and 127 represents
 silence) is obtained by the calculation given in appendix A of 
[RFC6465]: informally, level = -round(log10(audioLevel) * 20), with 
audioLevel 0.0 and values below 127 mapped to 127.

The non-standardized Chromium getStats has both `ssrc.audioInputLevel`
 and `ssrc.audioOutputLevel`. The source audio level would be the 
"input" audio level. Question is, should there also be a standardized 
stat for "output" level?

Please view or discuss this issue at 
https://github.com/w3c/webrtc-stats/issues/145 using your GitHub 
account
Received on Tuesday, 31 January 2017 16:06:40 UTC

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