Error resolution or example for webrtcdsp with webrtcechoprobe

[ankit]

Hello,

I am using calling functionality with rtp. But I am facing issue related to
echo when the call is ongoing.
To prevent this I added feature push to talk on temporary basis.
But to get rid of it, I found that webrtcdsp with webrtcechoprobe is a good
option.

So after integration in my C++ application when I run my app I am getting
errors as below:

GST::  "No echo probe with name webrtcechoprobe0 found."   164
GST::  "../../../gst-plugins-bad-1.12.4/ext/webrtcdsp/gstwebrtcdsp.cpp(523):
gst_webrtc_dsp_start (): /GstPipeline:audio_tx/GstWebrtcDsp:audio_rtc_dsp"

FYI : I have taken two gst elements for webrtcdsp & webrtcechoprobe but I
think when I am linking them at that time I am getting error as above. I am
using Gstreamer 1.12.4.

Please suggest any resolution or example, because on the forum there is not
any example related to webrtcdsp echo cancellation with c++.

Regards,
Ankit




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