> Just two general-purpose first order IIR sections
> is all you need for either the forward or inverse
> filter. Any textbook on digital filters will tell
> you how to program them. Inverting the LF filter
> requires an extra pole below the audio range to
> avoid infinite DC gain.
I'm aware of how to construct digital IIR filters :-) I was hoping
you had a URL to a nice official analog topology. The specific
implementation details matter.
> The channel EQ you'll find on most digital mixers is
> not linear-phase at all, nor acausal.
OK. Time to become incredibly overspecific:
Every digital EQ implementation I'm aware of for Linux is linear
phase. I wrote a few of 'em.
One could just build a digital equivalent of any of the old analog
topologies. For many filters (eg, compressors and the like) this is
totally the way to go. For EQ, I'll take a linear phase
implementation any day. That's the route taken by every piece of FOSS
EQ source code I've ever seen (it would not be surprising if I missed
a few). If you say VST has done a few this way (for whatever reason)
then I believe you.
> In almost all
> cases it's just first and second order IIR filters.
> For plugins anything goes, but the most of them are
> not linear phase, and no filter operating in real
> time can ever be acausal - by definition.
Negative delays are perfectly possible in digital. Well, if you
ignore the wallclock (assume a global system latency, and a local
negative latency within the system). It's just a semantic/terminology
argument at this point.
Cheers,
Monty
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Received on Sun Feb 14 08:15:02 2010
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