That’s a pretty deep question. You’ll need equations for shelving, band-pass/band-reject, low/high-pass filters and combine them into a parametric equalizer structure. Without working knowledge of DSP, you’ll find it very difficult, in my opinion.

This book would be extremely helpful: http://www.dafx.de/

If you have a very solid understand programming languages, you could perhaps learn the structures from the provided MATLAB code on the website. Personally, I would get some books, or do a ton of reading of academic papers, and study the equations within them to learn what equations you need.

Many equations are given in the frequency domain, which you will need to use the fft~ object to utilize.

fft(x[n]) = X(z)

H(z) = Y(z)/X(z)

x[n] is your audio signal, where n is each sample representing the audio signal. H(z) is the filter you are designing.

Very mathematical explanation:

http://www.collinsaudio.com/Prosound_Workshop/orfanidis%20decramping.pdf