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Re: IIR vs. FIR filters
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Re: IIR vs. FIR filters


  • Subject: Re: IIR vs. FIR filters
  • From: Michael Ashton <email@hidden>
  • Date: Wed, 11 Dec 2002 12:49:48 -0700

On Wednesday, December 11, 2002, at 03:03 AM, Urs Heckmann wrote:

Hi Michael,

just some thoughts:

Am Dienstag, 10.12.02, um 23:02 Uhr (Europe/Berlin) schrieb Michael Ashton:

Only one problem with FIRs though. Say you want to simulate a tunable Moog-style / 4-multiply filter. Isn't it easier to do that with an IIR? Perhaps you could do multi-rate processing of some sort through an FIR to get the tuning, and run a feedback loop for resonance .. dunno. I'll bet somebody's done this before.

Why would one try to implement an IIR by a FIR?

'Cause they're nuts. Like me. :)

(A resonant filter like the Moog cascade has an - almost - infinite impulse response)

FIRs are generally good at preserving phase, i.e. when seperating bands via windowed sinc kernel. However, they are expensive to compute (unless you use a moving average :-) Another drawback, if you need linear phase and want to compute efficently, you'll end up doing convolution by multiplication in the frequency domain (via FFT->multiply->iFFT), usually performing overlap-add, using double-sized kernels with zero padding, and adding not too little latency.

Yes, quite so.

Generally, as todays performance bottleneck is memory bandwidth, computational approaches should be preferred over memory wasting techniques. You'd rather use functional approximations to sin, cos, tan, pow than lookup tables.

Hmmmmm ... well, could be; although I'd venture to guess that, if a lookup table makes it into cache, it could be faster. However, the tremendous computation hardware on processors like the G4 might indeed make lookup tables a little pointless.

IIRs are fast to calculate, can easily be made time variant and stability problems can be met with waveshapers on the feedback (resonance). Latter is what often makes them sound better ("warmth", "drive", "fatness").

Now _there's_ an interesting thought. The Roland filter, which is one of the many variants on the gain-cell ARP design, has always been one of my favorites; and they always put a pair of diodes on the resonance feedback path to control the oscillation. I've not experimented much with different wave-shaping circuits on that path. Might be an interesting thing to do some time.

Plus, there are a lot of good explanations and source code on the net. I'd recommend oversampled Chamberlain derivates.

Hmm .. I've never heard of that. I'll look it up.

It occurred to me just now that a slick way to get around the biquad-warping problem for digital 4-multiply filters might be to take the impulse response of a continuous-time circuit and do convolutions ... hmm ...

You don't necessarily need a biquad filter. You could also cascade some 1-pole/1-zero filters or two 2-poles to get a 4-pole (if this is what you're looking for)

My sentence wasn't very well written - sorry. I wasn't thinking of biquad filters, but of the biquad transform frequently used to convert continuous-time filters into discrete-time filters. I think that, for music, the biquad transform doesn't work very well because it alters the shape of the transition band, which, in a music filter, is what produces a particular timbre.

Lots of memory though ..

Maybe I should bring this up on a computer-music list :)

Maybe check out this one:
www.musicdsp.org - mailing list, source code, advise. Nice people, not overly academic

Yes, that's the one I was thinking of.

(Watch out for "State Variable Filter, double sampled, stable" in the source code archive. This is a good one)

OK, 'preciate it!

thanks
----
Michael Ashton <email@hidden>
"Turning off Rendezvous to improve security is like having a company policy that every employee will be hit in the face with a baseball bat every day when they come to work in the morning, to discourage thieves."
-- Stuart Cheshire, the Rendezvous FAQ
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 >Re: IIR vs. FIR filters (From: Urs Heckmann <email@hidden>)

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