[EXT] Re: How to configure webrtc echo cancellation in Windows 10?

[Nicolas Dufresne]

Le vendredi 20 août 2021 à 11:51 +0000, Soebirk, Thorsten via gstreamer-devel a
écrit :
> Did you get this to work? I am facing a perhaps similar issue with a webrtc
> pipeline on Windows, where it seems like echo cancellation simply has no
> effect
> at all. In my pipeline I use directsoundsrc ! webrtcdsp ! ... ultimately going
> to webrtcbin and on the webrtc receiving side ... ! webrtcechoprobe !
> autoaudiosink buffer-time=40000 low-latency=true.

Try wasapisrc/sink with the low-latency=true property.

> 
> I have tried different values for buffer-time and low-latency, but it seems
> that
> regardless of what I do there is no audible echo cancellation at the other end
> of the webrtc connection.
> 
> I would appreciate any further advice.
> 
> Best regards 
> 
> Thorsten 
> 
> -----Original Message-----
> From: gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org> On
> Behalf
> Of Maksim Danilov
> Sent: 24. marts 2021 07:49
> To: gstreamer-devel at lists.freedesktop.org
> Subject: [EXT] Re: How to configure webrtc echo cancellation in Windows 10?
> 
> EXTERNAL MAIL: 
> 
> Thanks for the answer. I gave up on configuring AEC over rtp and came up with
> simple example that look like:
> wasapisrc ! audioconvert ! audio/x-raw, format=S16LE, rate=48000 !
> audioconvert ! webrtcdsp ! webrtcechoprobe ! audioconvert ! wasapisink. 
> It gives no result (in linux everything work as expected). 
> I tried you suggestion with low-latency mode, however got messages with
> 'invalid
> latency add some queues to the pipeline'. Even if I add queue to pipeline. I
> can't hear my self at all.
> That pipeline gives the same result: wasapisrc low-latency=true ! <queue> !
> wasapisink low-latency=true.
> So I think the problem is in plugin. Probably it add some additional latency
> in
> implementation, cause it doesn't act like real time.
> I inspected the code of plugin and it looks like a very fast implementation of
> api under Windows. Moreover it can't even capture specific caps of src/sink
> and
> uses shared format F32LE (maybe it is resampling and converting audio in core
> and we get some latency through).
> 
> 
> 
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