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Is it possible to do realtime noise-removal with PulseAudio, so the sound output that's coming out is smoother?

  • it probably is possible, though it would be intensive, and would require someone write a filter for it. – RolandiXor Dec 27 '10 at 12:36
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    @Vadi - I have added an answer that might help you if you are still using Ubuntu or any other PulseAudio based Distro. Hope it works for you buddy. – Luis Alvarado Apr 29 '16 at 21:23
  • This post seems to be doing something similar – akhmed May 10 '16 at 22:24
68

Pulseaudio module module-echo-cancel

I started reading a lot about PulseAudio and "hidden" options it had so I could find one that was similar to this question. The one I found was the noise-cancellation module, which is one that dramatically lowers any static noise on the microphone and even A LOT of the background noise, basically giving you the benefit of only recording your own voice with excellent quality (For audio recording for example). To do this follow this steps:

  1. sudo nano /etc/pulse/default.pa

  2. Add the following line anywhere on the file, but I recommend almost at the end where you will find a comment about Echo Cancellation stuff (~line 140):

    load-module module-echo-cancel
    
  3. Reload PulseAudio (pulseaudio -k) or simply restart the computer. You should be able to select the new Noise Cancellation option from the Input Device Section:

    screenshot

You can find more information about it on the Echo Cancel Module Page

Set input as default

If you wish to set as default the echo cancel device simply turn the above line into:

load-module module-echo-cancel source_name=logitechsource

and then at the bottom of the file add

set-default-source logitechsource

In this case I named the source logitechsource, but you can name it whatever you want and simply either restart pulseaudio.

Rename device

Lastly, if you do not want a super long name on the Sound Settings (When you want to select an input/output device). My suggestion is renaming the input device like this:

load-module module-echo-cancel source_name=logitechsource source_properties=device.description=LogitechHD

And again, restarting pulseaudio. The end result looks like this:

screenshot

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    Easy to do and doesn't seem to break things - thanks! – Vadim Peretokin Apr 30 '16 at 21:37
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    Excellent solution, worked out of the box for me, thank you – rancho Dec 28 '16 at 8:48
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    This cut most of the fan noise from my laptop and my recordings sound so much better! – PJ Brunet Nov 6 '17 at 6:49
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    Holy moly - I get amazing results even from a crappy $5 lapel microphone I got from eBay. It's not studio quality but it's perfectly fine for VOIP calls now. – Raymond Morgan Aug 27 '18 at 1:00
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    If you want to avoid the echo module from adjusting the volume slider automatically, you can set load-module module-echo-cancel aec_method=webrtc aec_args="analog_gain_control=0 digital_gain_control=1" as per gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/224 – joelostblom Oct 13 '18 at 23:08
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This is an old question, but I had the same problem and after some Googling (where I mostly found people who agreed it wasn't possible) and reading some man pages, I have now developed a solution based on user2330377's idea.

First you need to create a noise profile for SoX. Just use any audio recording program to record a few seconds of noise, then cd into the directory you saved it to and do sox noise.wav -n noiseprof noise.prof.

Then you need to create an ALSA loopback device:

sudo modprobe snd_aloop

This is required because pulseaudio, unlike Jack, cannot directly connect audio software together; we will hence use the loopback device as a proxy.

Now you need to start paman and find the names of both your microphone (or other recording device) and of the loopback device we just created. Once those are found, you can execute the following command to start recording sound from your microphone, piping it through SoX and then playing it on the loopback device:

pacat -r -d alsa_input.pci-0000_00_14.2.analog-stereo --latency=1msec|sox -b 16 -e signed -c 2 -r 44100 -t raw - -b 16 -e signed -c 2 -r 44100 -t raw - noisered noise.prof 0.2|pacat -p -d alsa_output.2.analog-stereo --latency=1msec

(Where you need to substitute the correct device names for the -d parameters -- the input device for the first pacat invocation and the loopback device output for the second.)

There you go, almost done! As a last step, start recording sound with the application of your choice, then start up pavucontrol, change to the "Recording" tab and set the audio device used for recording (displayed as the grey button to the right) to "Monitor of Loopback Audio Device". You should now have a clear and noise-free recording!

3

Investigation shows that there is no known way of doing real time noise reduction filtering with any Linux sub system. Some websites point to hardware you can buy which should do the trick much better than doing a software filter.

Alternatively if this is for a recording, you could pass the sound through Audacity and use the noise filter there.

  • Pity, I wonder why not. Thanks for investigating. – Vadim Peretokin Jan 8 '11 at 12:16
  • Sure! There is no any information on module documentation page about noise cancellation. There is only AEC (Acoustic Echo Cancellation) algorithm inside module-echo-cancel, which is have several implementations, like webrtc | speex. – vskubriev Feb 14 '18 at 8:16
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Here's at leas tone way, implement webrtc-aec in the

http://wiki.gentoo.org/wiki/PulseAudio

webrtc-aec Yes Uses the webrtc.org AudioProcessing library for enhancing VoIP calls greatly in applications that support it by performing acoustic echo cancellation, analog gain control, noise suppression and other processing.

Here's a paper dated 2013 on the subject (noise removal with pulseaudio not webrtc-aec specifically) http://lac.linuxaudio.org/2013/papers/37.pdf

"We have presented rst results of a multi-channel noise/echo reduction solution built on top of PulseAudio and motivated the design decisions. The work has resulted in a number of improvements in the PulseAudio echo cancellation and signal-processing framework, which have been contributed during the version 3.0/4.0 development cycle and should facilitate future embedded Linux audio solutions. Further work includes optimizing code for audio stream mixing, more ecient resampling methods, and the implementation of an efficient AEC in the multi-channel processing pipeline."

  • What I had in mind for a quick had was to takle a noise profile and do real time noise removal using sox and then play back the audio output which I can then use as an input stream in pulse. I believe that pulse is able to create and redirect input and output streams in this manner. That is the playback stream can get redirected as an input stream to another application and the playback stream is per application not per device, so it need not be routed to device hardware. – user2330377 Jun 22 '13 at 15:53
  • I'm giving this a go using pacmd to list pulseaudio devices, pacat to record and playback from them and sox to do the real time effects on the stream. – user2330377 Jun 22 '13 at 16:18
  • Would you be interested in posting the scripts for it? – Vadim Peretokin Jun 23 '13 at 20:51
0

There is no any information on module documentation page about noise cancellation. There is only AEC (Acoustic Echo Cancellation) algorithm inside module-echo-cancel, which is have several implementations, like webrtc | speex.

Therefore you should buy headset|microphones with integrated noise cancellation feature as much as possible.

As I tested voice recording in ubuntu I found some particular qualities:

  • Skype, Telegram uses raw input from the default device (in my case front-in-mic|backward-mic jacks). If you need to cancel a noise in this apps, you should buys headsets|microphones only with integrated noise cancellation feature

  • If you need to use voice calls in browser, then you should keep in mind that browsers has it's own voice processing algorithms implementations, e.g. WebRTC.

  • Also web applications (sites like talky, hangouts, appear.in, etc) can have it's own voice processing algorithms implementations, despite the fact that they can be based on webrtc hangout-analysis

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