noise suppression with librespeaker in RespeakerCorev2

Products & Technology ReSpeaker
#41

Hi Fred,



The capture and playback sample rate of seeed voicecard should be the same in Alsa setup. So we have to configure Alsa(edit /etc/asound.conf file) as follow:

</s>sudo nano /etc/asound.conf<e>

[code]pcm.!default {
type asym
#capture.pcm “hw:Loopback,1,0”
capture.pcm “seeedvoicecard”
#playback.pcm “plughw:seeed8micvoicec,1”
playback.pcm “play”
}

defaults.pcm.rate_converter “samplerate”

pcm.seeedvoicecard {
type plug
slave.pcm “hw:seeed8micvoicec,0”
#slave.rate 48000
#slave.format FLOAT_LE
ttable.0.0 1
ttable.1.1 1
ttable.2.2 1
ttable.3.3 1
ttable.4.4 1
ttable.5.5 1
ttable.6.6 1
ttable.7.7 1
}

pcm.play {
type plug
slave.pcm “hw:seeed8micvoicec,1”
slave.rate 48000
}[/code]

After configuration, you can still play your audio at any sample rate as you want, because Alsa will resample it to 48K automatically.

Hope this helps!



Thanks,

Jerry.

#42

Good catch!

That works great now.

Thank you very much



In the meantime, I also tried to play with the hybrid node but could not compile the line:

hybrid.reset(HybridNode::Create(true,3,1,0,true,0,false));



The compilation returned a load error:

$ g++ arecord_lv.cc -o arecord_lv -lrespeaker -lsndfile -fPIC -std=c++11 -fpermissive -I/usr/include/respeaker/ -DWEBRTC_LINUX -DWEBRTC_POSIX -DWEBRTC_NS_FLOAT -DWEBRTC_APM_DEBUG_DUMP=0 -DWEBRTC_INTELLIGIBILITY_ENHANCE

/tmp/ccPFm7fm.o: In function main':<br/> arecord_lv.cc:(.text+0x812): undefined reference to respeaker::HybridNode::Create(bool, int, int, int, bool, int, bool)’

collect2: error: ld returned 1 exit status



Do you know if this function is still in the new version of the library?



Thanks

Fred

#43

Hi Fred,



Yes, there is a bug in this node and thank you to point this out. I have recompiled the librespeaker, please download it again at this link:

<LINK_TEXT text=“https://v2.fangcloud.com/share/11a20219 … f8?lang=en”>https://v2.fangcloud.com/share/11a20219f73655959b42aa57f8?lang=en</LINK_TEXT>



Thanks so much!

Jerry

#44

This one works perfectly well

Thanks you very much

Great service

Fred

#45

Hello,



I have some issues with with the noise reduction example using aloop.

When I record after an “aplay” command, I have a big initial pop in my recording.

For example:

aplay phrase.wav

sox -d -t wavpcm -c 1 -b 16 -r 16000 -e signed-integer --endian little record.wav trim 0 1

Then in my file record.wav, there is a big pop at the beginning that you can hear with aplay.



If I record twice in a row, then the second file is good, meaning:

aplay phrase.wav

sox -d -t wavpcm -c 1 -b 16 -r 16000 -e signed-integer --endian little record.wav trim 0 1

sox -d -t wavpcm -c 1 -b 16 -r 16000 -e signed-integer --endian little record2.wav trim 0 1

The file record.was has the pop but the file record2.wav is good (no initial pop in it).



I hope that you can reproduce and let me know how to fix this because it is causing problem when submitted next to the ASR.



Moreover, I noticed big variation in the volume of my recording. It varies from one recording to another but sometimes within the recording itself (e.g. starting loud and finishing low). I played with the agc parameter and even disable it but without any luck.



Thanks for your help

#46

Hi Fred,



Can you try to record at first and aplay after it? And the format of pharse.wav should be same as the recordings, which means it should be 16-bit. If pharse.wav is not 16-bit, I think you should use an Alsa plugin to play it.

#47

Hi,



i just came across this topic, is there any progress on this project?

And is it possible to share the beta image again, since i would like to test and help :slight_smile:



Thanks in advance!



Best regrads,

Peter