Many applications that produce sound, such as software synthesizers, are able to apply a filter that varies with time, such as applying a low pass filter that varies with an LFO.

I currently have a means of calculating coefficients for an IIR filter given poles and zeros, but do not know how to achieve this varying without redoing the calculations from scratch each time it needs to change. Is there a good approach to this that does not rely on using only a first order filter?

  • $\begingroup$ oh, this is a good question. are you making a 4-pole Moog filter or something like it? $\endgroup$ – robert bristow-johnson Aug 30 '19 at 3:31
  • $\begingroup$ I'm actually trying to work on a more flexible sound generating program/library in general, with many potential functions to it, including filters (perhaps all-pole) that can vary their peak frequency over time to achieve effects like formant transitions. $\endgroup$ – user2649681 Aug 30 '19 at 11:48

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