Gstreamer echo canceller udpsrc and udpsink

[Shrikant Lahase]

I am getting confused where to keep element *"webrtcprobe" & *
*"webrtcechoprobe" . i tried but not getting result.*
Can you please tell me exact location in my commands:

Recording & udpsinkcommand:


*"gst-launch-1.0 -v alsasrc ! audioresample ! audioconvert ! audio/x-raw,
format=S16LE, layout=interleaved, rate=8000, channels=2 !  udpsink
host=192.168.1.139  port=5001"*
 Receive & playback command:



*"gst-launch-1.0 -v --gst-debug-level=4  udpsrc port=5003 !  audio/x-raw,
layout=interleaved, rate=8000, format=S16LE, channels=2 ! audioconvert !
autoaudiosink" *

*Thanks,*

*Shrikant*

On Thu, Sep 14, 2017 at 4:28 PM, Shrikant Lahase <shrikant at neotechindia.com>
wrote:

> 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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