Gstreamer echo canceller udpsrc and udpsink

[Shrikant Lahase]

Thanks for your reply.

For using *"webrtcdsp" & "webrtcprobe"* which plugin needs to install?
There is one more option I read on google
*"webrtcechoprobe"  *
What is difference between  *"webrtcprobe" & **"webrtcechoprobe"  *

What is attached signature.asc file?


On Wed, Sep 13, 2017 at 11:27 PM, Nicolas Dufresne <nicolas at ndufresne.ca>
wrote:

> 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
> > _______________________________________________
> > gstreamer-devel mailing list
> > gstreamer-devel at lists.freedesktop.org
> > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20170914/25d02ce6/attachment-0001.html>