Re: WebRTC and AGC

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noa.gr...@gmail.com

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Feb 17, 2013, 6:48:04 AM2/17/13
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Hi,
Can you please explain when do you recommend to use Adaptive analog and when Adaptive digital? It seems that no matter what Im using still the audio Im sending is too load as compare to the others
Noa

On Wednesday, February 13, 2013 9:38:53 AM UTC+2, noa.gr...@gmail.com wrote:
Hi,
We are using webRTC version:1869
The parameters that we are using for the AGC is:
Automatic, Target Level:10, Compression Gain:8, limiter=true.
It seem that the output of the Audio is too load in compare to the others in the calls (difference of 20db)
We tried to change it to Adaptive Digital,Target Level:16, Compression Gain:16, limiter=true.
It did reduce few db but not enough.
Are you aware of such issue? Do you have any other suggestions of configuration?

Henrik Andreasson

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Feb 18, 2013, 3:10:47 AM2/18/13
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Adaptive digital is mainly intended for devices which does not have an additional analog (slider), e.g. for mobile devices. Adaptive analog is intended for desktop platforms. Adaptive analog contains a digital component as well.

Usage of the additional parameters requires deep knowledge about a particular hardware and is not intended as default settings.

We do not support the native WebRTC components but only the JS level APIs which uses them. Do let us know if you have issues in e.g. Chrome or Firefox with any of these components.

Thanks


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noa.gr...@gmail.com

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Feb 18, 2013, 7:53:33 AM2/18/13
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We are using desktop platform and we do set adaptive analog to the webRTC but the audio going out from this application is too loud as compare to the others. It seems that the AGC doesnt work as we do expect.
Any suggestion what to do? How to lower the Audio?

Henrik Andreasson

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Feb 18, 2013, 8:06:58 AM2/18/13
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As mentioned above, we are unable to support all possible integrations of the native APIs. Can you try using apprtc.appspot.com in Chrome and verify that the AGC works there? Check the volume slider of the selected default mic and verify that it adjusts as it should speak into the mic. If it does not, please file an issue report in the WebRTC issue tracker.


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