Gstreamer echo canceller udpsrc and udpsink

[Shrikant Lahase]

Is GStreamer  1.12 version is necessary for echo cancellation?

On Thu, Sep 14, 2017 at 9:49 AM, Shrikant Lahase <shrikant at neotechindia.com>
wrote:

> 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-fo
>> r-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
>
>
>
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