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Old 01-19-2013, 03:47 PM   #1
matters
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removing mic noise problem


Im having trouble removing microphone backround noise. By executing arecord only it shows:

Code:
arecord test.wav
Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
^CAborted by signal Interrupt...
however bacground noise is fixed by changing the sample rate - by executing:

Code:
arecord -f dat -r 48000 -D hw:0,0 -d 5 test.wav 
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
is it anyhow possible to change it so it will be changed by default to 48000 via alsamixer? thanks!

thanks
 
Old 01-20-2013, 05:48 AM   #2
elesmod
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I don't know if you can set it in alsamixer (you'll have to wait for people with more knowledge to respond), but I might have a different solution for you.

Make an alias, so that the command 'arecord' will do something different. In the terminal type this:
Code:
alias arecord='arecord -f dat -r 48000 -D hw:0,0 -d 5'
Note that after reboot, the alias will be gone. If you want it to stick through restarts, add that line to your .bashrc and/or (depending on how you're loging into your terminal) .bash_profile.

Last edited by elesmod; 01-20-2013 at 05:50 AM.
 
Old 01-20-2013, 06:00 AM   #3
ukiuki
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Converting Sample Rates On Input
This is probably what you need.

Regards
 
Old 01-21-2013, 04:31 AM   #4
matters
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Quote:
Originally Posted by ukiuki View Post
Converting Sample Rates On Input
This is probably what you need.

Regards
Thats what ive tried before i posted this thread. I have created .asoundrc in the user home directory, and .asoundrc contained only the following lines:

Code:
pcm.rate_convert {
    type plug
    slave {
        pcm "hw:0,0"
        rate 48000
    }
}
That however, dosent solve the problem. Do i have to add something more to the file in order to make it work?

Btw, arecord parameters was just an example of how changing sample rate to 4800Hz fixed the problem. I need mic to work especially with skype.
Thanks!
 
Old 01-21-2013, 05:09 AM   #5
dh_
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Are you sure it's the modified sampling rate that fixes it? I have a similar issue and it seems to be the “-f dat” flag which fixes it.
 
Old 01-21-2013, 10:49 AM   #6
matters
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Quote:
Originally Posted by dh_ View Post
Are you sure it's the modified sampling rate that fixes it? I have a similar issue and it seems to be the “-f dat” flag which fixes it.
From my end -f dat <filename.wav>, although it shows that specifies rate by default to 48000 it dosent fix the problem, it fixes it only when i explicitly changed the sample rate, just like i presented above.
 
  


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