> 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 _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
