🚀 go-pugleaf

Okay, here's a question for any of you digital filter types...

What is the best continuous-time to discrete-time transform to use
when designing a digital filter for audio signals?  I have done some
work with discrete time control systems and am familiar with a few
different methods:  impulse invariant, step invariant, bilinear (with
and without freq. prewarping), and pole-zero mapping.  Are any of these
methods commonly used for audio filter design?  Does it depend on the
desired frequency response characteristics of the filter?

If I sample at 44.1 KHz, will 20 KHz frequencies be a problem
(since it is so close to the folding frequency of 22.05 KHz)?  Would
one of the above transform methods handle the problem better than the 
others.  I was considering using either pole-zero mapping (since it
would map 0 Hz - 22.05 KHz to 0 Hz - infinity, which might not be bad
for my application), or the bilinear transform with frequency prewarping
(because the name sounds so cool), but since I've never even read a book on
discrete filter design (perhaps I should, no?) I thought I'd get opinions
from people who have a clue first.

Any opinions, ideas, comments (or, yes, even references to good texts)
would be most appreciated.

Thanks...

-Kirk

How 'bout them Raiders?

--
      Kirk Norton
      16399 West Bernardo Drive                                  
      San Diego, CA  92127-1899