Path: utzoo!utgpu!news-server.csri.toronto.edu!cs.utexas.edu!usc!wuarchive!emory!rsiatl!larry
From: larry@rsiatl.UUCP (Larry Kahhan)
Newsgroups: comp.dsp
Subject: Re: Digital Filter Design
Message-ID: <4900@rsiatl.UUCP>
Date: 15 Nov 90 23:11:12 GMT
References: <1990Nov14.194941.16247@watserv1.waterloo.edu>
Reply-To: larry@rsiatl.UUCP (Larry Kahhan)
Distribution: na
Organization: Rapid Deployment Systems, Inc. (making go fast things and things go fast)
Lines: 32

In article <1990Nov14.194941.16247@watserv1.waterloo.edu> kevo@audiolab.uwaterloo.ca (Kevin Kotorynski) writes:
>I am trying to implement some COMPLEX transfer functions which 
>are not minimum-phase, and I need to retain the magnitude AND
>PHASE characteristics of the original filter transfer function.
>
>The impulse responses are too long for realization using FIR filters,
>since real-time processing is required.  Can any-one suggest an
>algorythm or some good reading to help me predict the
>coefficients for a direct-form filter, given the reqired impulse
>response or transfer function?  Any suggestions welcome; please
>E-Mail or Post responses.


MathCad will do what you want. I've used it to fit arbitray transfer
functions representing IIR type filters against a complex frequency
response dataset. This is probably the simplest approach.

MatLab has similar capabilities using their System Identification
Toolbox.

Another approach would be to use an adaptive filter and let it find
its own coefficients.

If you're into Fortran, there is a program in the IEEE DSP collection
by Andy Deczky that will also do what you want.


In sum, it pretty much depends on which tools you happen to have
available. I like the MathCad approach, but that's my personal
preference.

Larry Kahhan - NRA, NRA-ILA, CSG, GSSA , & GOA