I have a couple audio programming books (Zolzer DAFX and Pirkle Designing Audio Effect Plugins in C++). All the filters they describe were easy enough to program.
However, they don't discuss having the frequency and resonance (or whatever inputs a given filter has--parametric EQ etc.) CHANGE. I am doing the expensive thing of recalculating all the coefficients every sample, but that uses a lot of CPU. My questions are: 1. Is there a cheaper way to do this? For instance can one pre-calculate a big matrix of filter coefficients, say 128 cutoffs (about enough for each semitone of human hearing) and maybe 10 resonances, and simply interpolating between them? Does that even work? 2. when filter coefficients change, are the t-1 and t-2 values in the pipeline still good to use? I am using them and it SEEMS fine but now and then the filters in rare cases go to infinity (maybe fast changes with high resonance?) and I wonder if this is the cause. 3. Would you guess that most commercial software is using SIMD or GPU for this nowadays? Can anyone confirm at least some implementations use SIMD or GPU? Frank
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