[linux-audio-dev] applying RIAA curves in software

[fons adriaensen]

On Sat, Oct 29, 2005 at 03:10:50PM +0300, Jussi Laako wrote:

> On Wed, 2005-10-26 at 02:41 +0200, fons adriaensen wrote:
>
> > Filter 1:   F =   50 Hz, A = 9
> > Filter 2:   F = 2120 Hz, A = 1
> > 
> > and add the two outputs.
> 
> >From quality point of view, at least I would recommend using IIR filters
> for this...

Could you explain ? The (trivially simple) first order lowpass sections
will have the correct amplitude _and_ phase response. 

> Unless digital'ish sound is preferred... ;)

What the **** is "digital'ish" sound, and how is it relevant here ?

-- 
FA