Gstreamer echo canceller udpsrc and udpsink

[Nicolas Dufresne]

Le mercredi 13 septembre 2017 à 22:36 +0530, Shrikant Lahase a écrit :
> Hi

Welcome to the mailings list.

> 
> I need to implement echo canceller in my audio application.
> I am sending audio alsasrc to remote mobile android app from my hardware.
> The command is as follows:
> 
> Sending and receive combined in single command:
> 
> "gst-launch-1.0 -v alsasrc ! audioresample ! audioconvert ! audio/x-
> raw, format=S16LE, layout=interleaved, rate=8000, channels=2,
> endianness=4321, width=16, depth=16, signed=true ! udpsink host=$1
> port=5001 & gst-launch-1.0 -v --gst-debug-level=4  udpsrc port=5003 !
> audio/x-raw, layout=interleaved, rate=8000, format=S16LE, channels=2,
> endianness=4321, width=16, depth=16, signed=true ! audioconvert !
> autoaudiosink"
> 

This was very recently asked on the list,

http://gstreamer-devel.966125.n4.nabble.com/Webrtc-plugin-for-AEC-support-td4684580.html

In this thread, I give a example pipeline for testing in loopback the
AEC. Note that you are using 0.10 syntax in your pipeline.

  audio/x-raw, layout=interleaved, rate=8000, format=S16LE, channels=2,endianness=4321, width=16, depth=16, signed=true

Is in fact:
  audio/x-raw,layout=interleaved,rate=8000,format=S16LE,channels=2

depth,width,signed,endianess have no meaning in 1.0 and may lead to
negotiation error. The in-gstreamer echo canceller works by placing
webrtcprobe close to your load speaker playback element, and placing
webrtcdsp filter close to your recording element.

Note that you'll always get better result due to timing precision with
using AEC in Pulse Audio rather then in Gst.

> 
> But right now I am doing push to talk to avoid echo and noise on the hardware side.
> 
> Please tell me how to implement AEC in GStreamer.....
> 
> 
> Thanks,
> Shrikant
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