Hi,
I tried to search the posts, but could not find the right answer for my problems that Im facing with audio tuning.
We are creating a product where one peer is native WebRTC application on Android platform and other end is web app running in Chrome.
Everything is working quite nicely, but I have some audio quality issues, especially in Automatic Echo Cancellation (or so I think). The following is my
common man analysis.
* There is too much echo cancellation
- The audio is not clear enough, it seems that it starts to eat it's own signal. Audio becomes "Wobbly", sometimes you hear what the other person
is saying and sometimes you need to ask them to repeat what they just said.
If I remove completely AEC, the audio is much more clearer (quite much), not "wobbly" anymore, but I introduce echo to the call, which is not OK.
So my question is how I can tune the AEC? Is it even possible? And how are the following parameters interpreted (from: chrome://webrtc-internals):
Are the *EchoCancellation* parameters "tuneable", or is AEC like on/off switch? If those are tuneable, how are the parameters used. Any help is welcome. Thanks!
-Mikael
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| googEchoCancellationQualityMin | 1 |
| googEchoCancellationEchoDelayMedian | -60 |
| googEchoCancellationEchoDelayStdDev | 0 |
| googEchoCancellationReturnLoss | -100 |
| googEchoCancellationReturnLossEnhancement | -100 |
| googCodecName | opus |
| googTypingNoiseState | false |
| audioOutputLevel | 38 |
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