How to implement an Equalizer
I know there are a lot of questions about equalizers in so, but I didn't get what I was looking. What I want to do is an equalizer for modifying audio samples in such a way like:
equalizer.eqAudio(audiosamples, band, gain)
I'm not sure if that is the exact interface that I want because I know little about DSP in terms of implementing them (I used filters, limiters, compressors but not made them).
So Googling about this I read that I must do a FFT to the samples so I get the data per frequency ranges instead of amplitude, process it the way I want and then make the inverse of the FFT so I get the result in audio samples again. I looked for an implementation of this FFT and found JTransform for Java. This library has an implementation of a FFT related algorithm called Discrete Cosine Transform (DCT).
My questions are :
- Well, Am I in the right way?
- Since FFT gives me data about frequency, I should pass to the FFT algorithm a chunk of samples. How big this chunk must be?
- Is there a good book about DSP programming that explains equalizers ?
Thanks!
Answer
FFT wouldn't be my first choice for audio equalisation. I would default to building an EQ with IIR or FIR filters. FFT might be useful for special circumstances.
A commonly recommended reference is the Cookbook Formulae for Audio EQ Biquad Filter Coefficients.
A java tutorial for programming biquad filters. http://arachnoid.com/BiQuadDesigner/index.html
Is there a good book about DSP programming that explains equalizers ?
Understanding Digital Signal Processing is a good introduction to DSP. There are chapters on FIR and IIR filters.
Intoduction To Digital Filters with Audio Applications by Julius O. Smith III.
Graphic Equalizer Design Using Higher-Order Recursive Filters by Martin Holters and Udo Zolzer is a short paper detailing one EQ filter design approach.