webrtcdsp and webrtcechoprobe on Windows
nicolas at ndufresne.ca
Wed Jun 24 20:21:22 UTC 2020
Le mar. 23 juin 2020 21 h 30, Attila <kocsisa01 at gmail.com> a écrit :
> I've tried setting the latency-time to a value lower than the buffer-time.
> There's a small improvement, but the problem with the ringing persists.
> I've recorded the result and shared the file (echo.mp3) on my google drive:
> To make it work, I had to increase the buffer-time on both the source and
> the sink. I've also tried adjusting the noise-suppression-level and
> echo-suppression-level, but setting them both to high seems to give the
> result (no change).
> After some testing, the pipeline below seems to give the best result.
> gst-launch-1.0 -v -e wasapisrc buffer-time=600000 ! audioconvert !
> audio/x-raw, layout=non-interleaved ! webrtcdsp
> echo-suppression-level=high ! audioconvert ! webrtcechoprobe ! audioconvert
> ! wasapisink buffer-time=20000 latency-time=15000
That is strange, but I'm not familiar with the wasapi implementation.
Normally you use 20/10 or 30/10, basically buffer-time is usually a
multiple of latency-time.
> At this point, I'm wondering how I should proceed... I can't seem to get
> this to a point where a 2-way conversation without headphones can be
> out and it seems unlikely, that I can fix this by following up on short
> hints (which are greatly appreciated!).
The fact you still get ringing might indicate inaccurate latency reporting
in this plugin. The accuracy could in fact change depending on the HW
driver behind. Have you tried using delay agnostic mode ?
> Is there a way to get someone to give me a little more hands on assistance?
> If so, how should I go about that?
> Am I even on the right track trying to do this with gstreamer? ...or should
> I try coding this using the relevant libraries directly (clearly much more
> work, but if it works better)?
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
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