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.
Regards,
Ankit
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