Error resolution or example for webrtcdsp with webrtcechoprobe
nicolas at ndufresne.ca
Tue Mar 19 14:13:28 UTC 2019
Le mar. 19 mars 2019 06 h 11, ankit <ankit.patel1 at einfochips.com> a écrit :
> 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
> 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_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++.
Have you read:
Appart couple of cast, C or C++ should be identical. Make sure you
understand why there is two elements and what's their purpose. You cannot
randomly place these in your pipeline.
If you are on Linux with Pulseaudio, I strongly recommend using the
PulseAudio integration, as the probe and cancellation is a) in a real time
thread and b) much closer to were we communicate with the hardware.
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
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