Webrtc plugin for AEC support

[be999]

Hi All,

I am trying to build a 2-way audio application with AEC.

A simple test application like following transports a sine wave over udp:
gst-launch-1.0 audiotestsrc \
! audio/x-raw, format=S16LE, layout=interleaved, rate=48000, channels=1 \
! udpsink host=127.0.0.1 port=5006 sync=false \
udpsrc port=5006 \
! audio/x-raw, format=S16LE, layout=interleaved, rate=48000, channels=1 \
! queue ! alsasink sync=false

As soon as I add the webrtc plugins in the pipeline, the source branch does
not produce any packets anymore 
GST_DEBUG=*udp*:6 gst-launch-1.0 audiotestsrc \
! audio/x-raw, format=S16LE, layout=interleaved, rate=16000, channels=1 \
! webrtcdsp \
! udpsink host=127.0.0.1 port=5006 sync=false \
udpsrc port=5006 \
! audio/x-raw, format=S16LE, layout=interleaved, rate=16000, channels=1 \
! webrtcechoprobe \
! alsasink sync=false

Any idea why this is happening?

Thanks!



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