Algorithms for sound filters, like reverb, dynamic range compression, lowpass, highpass, notch, etc.

It's easy to find countless math equations on sound filters, but a bit harder to find simple source code. This library is my attempt at cleaning up and presenting the math-heavy filter algorithms for the programming community.

Please note that I favored simple code over fast code. Hopefully it's made it more understandable.

(MIT License)

The ./build script is a simple bash script that compiles the source files using clang. It's dirt simple, I promise.

Simply run ./build and the executable should be ./tgt/sndfilter.

• Reverb (Algorithmic)
• Compressor
• Low-Pass (Cutoff, Resonance)
• High-Pass (Cutoff, Resonance)
• Band-Pass (Frequency, Q)
• Notch/Band-stop (Frequency, Q)
• Peaking (Frequency, Q, Gain)
• All-Pass (Frequency, Q)
• Low Shelf (Frequency, Q, Gain)
• High Shelf (Frequency, Q, Gain)

The reverb.c, compressor.c, and biquad.c are the core algorithms.

I do not understand the biquad math, so please don't ask me any questions :-). The core formulas were extracted from Biquad.cpp (Chromium source), and cleaned up a bit to make easier to read.

The compressor came from DynamicsCompressorKernel.cpp (also from Chromium), and cleaned up a bit more. I swapped out the adaptive release curve and simplified the knee calculations. I feel a little more comfortable with that algorithm because there isn't a whole lot of magical math involved.

The reverb effect is a complete rewrite of Freeverb3's Progenitor2 algorithm. It took quite a lot of effort to tear apart the algorithm and rebuild it, but I'm pretty sure it's right.