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[webrtc-stats] Stat for how many audio stream packets are expanded when the user is speaking

From: henbos via GitHub <sysbot+gh@w3.org>
Date: Wed, 01 Feb 2017 10:34:04 +0000
To: public-webrtc@w3.org
Message-ID: <issues.opened-204547682-1485945242-sysbot+gh@w3.org>
henbos has just created a new issue for 
https://github.com/w3c/webrtc-stats:

== Stat for how many audio stream packets are expanded when the user 
is speaking ==
The non-standardized Chromium getStats contains the following stat:
```
// Fraction of packets that are expanded (synthesized) when we detect
// that the user is speaking.
ssrc.googSpeechExpandRate
```
What exactly does this mean? Is this useful? Should we standardize it 
or something similar? Sums and counters are preferred over rates.
RTCInboundRTPStreamStats already have various packet counters. Should 
we add another `unsigned long packetsSpeechExpanded`?


Please view or discuss this issue at 
https://github.com/w3c/webrtc-stats/issues/152 using your GitHub 
account
Received on Wednesday, 1 February 2017 10:34:11 UTC

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