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[webrtc-stats] RTCSenderAudioTrackAttachmentStats: Audio Device Errors (#429)

From: Bernard Aboba via GitHub <sysbot+gh@w3.org>
Date: Mon, 15 Apr 2019 19:33:17 +0000
To: public-webrtc-logs@w3.org
Message-ID: <issues.opened-433443749-1555356796-sysbot+gh@w3.org>
aboba has just created a new issue for https://github.com/w3c/webrtc-stats:

== RTCSenderAudioTrackAttachmentStats: Audio Device Errors ==
Currently we do not keep statistics relating to audio device errors or the audio device that was the source of a track. 

Audio device errors can have a significant impact on audio quality. While information relating to an audio device (e.g. a microphone or speaker) is available in `track.label`, this information is not tracked in stats, nor are device failure reasons that surface at initialization and mid-call, including: 

- Already initialized,
- Unsupported format,  
- Buffer too large,
- Device stall,  
- No interface,  
- Device hung, 
- Exclusive mode only 

Other potentially useful information includes: 

o       Percentage of call duration that the speaker or microphone was muted (at a lower level than app where the app is unaware)
o       Master (and where applicable, session) volume level for render and capture (if analog AGC is not enabled)

Device information which may be helpful for diagnostics purposes (but which might represent additional fingerprinting surface) include:

o       PID/VID of mic and spk device in use
o       Device form factor (headset / speaker phone / etc.)
o       Device interface (USB/BT/PCI/etc)






Please view or discuss this issue at https://github.com/w3c/webrtc-stats/issues/429 using your GitHub account
Received on Monday, 15 April 2019 19:33:18 UTC

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