You are supposed to be able to define the master source & sink
My usual is
>>>load-module module-echo-cancel use_master_format=1 aec_method=webrtc aec_args=“analog_gain_control=0 digital_gain_control=1 agc_start_volume=85” source_name=echoCancel_source sink_name=echoCancel_sink
>>>
>>>set-default-source echoCancel_source
>>>set-default-sink echoCancel_sink
use_master_format=1 just seemed easier than the problems I had in attaining a working defined sink/source
I have forgot where I found the details for master_sink & master_source parameters maybe its in the code but going to have a good look again at PA webrtc.
I also got into some problems as wanted to have asound.conf that would use pulse.
Using pulseaudio_alsa and
>>># Use PulseAudio by default
>>>pcm.!default {
>>>type pulse
>>>fallback "sysdefault"
>>>hint {
>>> show on
>>> description "Default ALSA Output (currently PulseAudio Sound Server)"
>>>}
>>>}
>>>
>>>ctl.!default {
>>> type pulse
>>> fallback "sysdefault"
>>>}
But seemed to get stuck in this tail chasing of default sound on reboots.
As said this weekend going to have another good look at PA Webrtc AEC on the Pi but I found when using the Pi output and usb mic the clock drift was quite excessive.
I am using soundcards with a singular clock and card for sink/source and getting similar results which has me head scratching.
You can wire up 2x I2s mics to a Pi quite easily and that would share the same clock as the Pi output (doesn’t work pi4 sadly)
I will look forward to how you get on as my SpeexEc results have been quite good on the Pi and expected PA WebRTC AEC to be better but actually seems worse.
PS both of us are setting autogain and from speex that isn’t really a good idea as it makes thing much harder for the adaptive FFT routine.
AGC needs really long attack/delay values in fact with voice in manner of speech its hard to use AGC and not just amplify noise.
Really the gain shouldn’t be there are its problematic anyway.
Stuart
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