[pulseaudio-discuss] Microphone DC-Offset compensation and noise filtering.

[IL'dar AKHmetgaleev]

На Mon, 10 Jan 2011 11:44:36 +0100
Maarten Bosmans <mkbosmans at gmail.com> записано:

> 2011/1/10 IL'dar AKHmetgaleev <akhilman at gmail.com>:
> > I have a cheap USB headset which records audio with the little
> > DC-Offset and noise at 50Hz.
> >
> > As I know it's very usual problem of cheap audio devices and
> > integrated audio cards.
> >
> > So I'm requesting a module which will filter input stream.
> > This pseudo code was suggested in alsa ML:
> >  
> >> xs=0
> >> while (input){
> >> xs=.01 xinput +.99 xs
> >> xoutput=xinput-xs
> >> }
> >> (This averages over roughly the last 100 inputs and subtracts the
> >> offset). If you want a longer averaging, change the coefficients.  
> 
> It is wrong to say that the averaging is done over the last 100
> inputs. What you are proposing is averaging with exponential
> weighting. This means the averaging window is infinately long.
> 
> The concept can be useful, but you need a much smaller factor if by
> inputs you mean samples. 0.00001 for example would be better, because
> then the last period of a 20Hz wave in a 48000Hz signal only
> contributes 2.4% to the average (1-.99999^2400), which seems much more
> reasonable.

You know better how to do it. It's why I'm asking here.

> > Will be nice to have such module with sampling period as
> > attribute.  
> 
> Is there any reason this kind of signal processing wouldn't be more
> appropriate in an alsa driver?

They suggested to buy correct hardware on mailing list. ;)