On 12/01/2020 5:06 PM, Frank Sheeran 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?
It depends on the filter topology. Coefficient space is not the same as
linear frequency or resonance space. Interpolating in coefficient space
may or may not produce the desired results -- but like any other
interpolation situation, the more pre-computed points that you have, the
closer you get to the original function. One question that you need to
resolve is whether all of the interpolated coefficient sets produce
stable filters (e.g.. keep all the poles inside the unit circle).
2. when filter coefficients change, are the t-1 and t-2 values in the
pipeline still good to use?
Really there are two questions:
- Are the filter states still valid after coefficient change (Not in
general)
- Is the filter unconditionally stable if you change the components at
audio rate (maybe)
To some extent it depends how frequently you intend to update the
coefficients. Jean Laroche's paper is the one to read for an
introduction "On the stability of time-varying recursive filters".
There is a more recent DAFx paper that adresses the stability of the
trapezoidally integrated SVF. See the references linked here:
http://www.rossbencina.com/code/time-varying-bibo-stability-analysis-of-trapezoidal-integrated-optimised-svf-v2
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?
I don't have an answer for this, by my guesses are that most commercial
audio software doesn't use GPU, and that data parallelism (GPU and/or
SIMD) is not very helpful to evaluate single IIR filters since there are
tight data dependencies between each iteration of the filter. Multiple
independent channels could be evaluated in parallel of course.
Ross.
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