Hi Frank—As far as I know, GPU processing of audio in plug-ins is pretty rare, and I don’t know that it would be a great fit for filters, but you can recall coefficients as needed (as Davide pointed out), and interpolation between points my be OK, it would take some analysis and dependent on the interpolation accuracy…mainly I wanted to say you might try kvraudio DSP forum, where practical synth filters have been discussed quite a bit over the years.
> On Jan 11, 2020, at 10:06 PM, Frank Sheeran <fshee...@gmail.com> wrote: > > 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 _______________________________________________ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
