Error resolution or example for webrtcdsp with webrtcechoprobe

[Nicolas Dufresne]

Le mercredi 20 mars 2019 à 01:40 -0500, ankit a écrit :
> Hello Nicolas,
> 
> Thanks for reply.
> 
> I understand your point and explanation about echo cancel.
> 
> Currently I am using below reference pipeline:
> 
> gst-launch-1.0 *alsasrc* !
> audio/x-raw,format=\(string\)S16LE,rate=16000,channels=1 ! *webrtcdsp* !
> audioconvert ! speexenc ! rtpspeexpay pt=111 ! udpsink host=10.102.99.51
> port=7078 *udpsrc* port=7078
> caps="application/x-rtp,media=\(string\)audio,clock-rate=16000,encoding-name=\(string\)SPEEX,encoding-params=\(string\)1,payload=111"
> ! rtpspeexdepay ! speexdec ! audioconvert ! *webrtcechoprobe* ! alsasink
> 
> With the above pipeline however I am not able to remove echo, also I need to
> restart board to use it again(no error occurred while pipeline is running).
> If I remove webrtc component from pipeline then it works without reboot.

alsasink default latency is configured to 200ms, this is way too high
for the webrtcdsp algorithm. Try reducing it (buffer-time) to something
like 80ms. The algo works best if the probe ss close in term of delays
to the playback (hence my pulseaudio comment).

> 
> This same pipeline I have implemented with c++ code so without webrtc
> element it works properly but when I try to add that element I am getting
> error as mentioned before in my first post.
> 
> I hope you can understand my issue.
> 
> Regards,
> Ankit
> 
> 
> 
> --
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