Filtering and Distorting Audio stored in byte array Java

Elements needed:

• read bytes (through AudioInputStream probably, if data isn't being held in a memory array)
• convert audio data bytes to PCM
• oversample
• apply the distortion function
• anti-alias filtering
• decimate
• convert back to bytes
• output via SourceDataLine

I don't know what the best distortion algorithm might be for your purposes. There are a number of suggestions in DSP Related's free book Physical Audio Signal Processing. I recall that Stanford University's CCRMA had some interesting articles on this as well, but I don't have the links handy.

My personal favorite distortion is the tanh function. The Java code that I use for a synthesized patch that gets a fairly thick guitar-like distortion follows:

float distortVal = (float)Math.tanh(noteVal * (1 + timbreNormal * 5));

In this case, noteVal is a single PCM value. I have a variable timbreNormal ranging from 0 to 1, and am reading it from the equivalent of a foot pedal controller. The "magic number" anti-pattern value of 5 is a constant that I lazily put in. It determines the maximum amount by which the signal is amplified before applying the distortion algorithm. Whatever distortion algorithm you are using, the intensity of the distortion will depend on the degree to which you amplify the signal before applying the algorithm. If you want a more subtle effect, the "magic number" value could be set lower.

Oversampling helps a huge amount with the filtering step. I oversampled by 8, and am using a two-pole Butterworth filter. I'd have to do some significant digging to find the values used in the filter design. But I think you can get the gist of the program flow from this answer.