Hi all,
For the past few weeks I have been working on a custom type reson comb filter external. Using SndObj dsp library I have two comb filter( one on each channel left and right). There is a low pass filter placed in serial after each comb. The effect til now is working quite well. But I think my understanding of the freq response of the comb filter is a little hazy hence its implementation a little naive. My understanding is that the freq response is determined by the position of the Poles on the unit circle. L the number of samples delay (in sample time => t ( in secs) / the sampling rate) also denotes the number of poles there will be.
Up until now I have used a simple calculation to convert the freq response in hertz to the number of samples delay (L).
float L = Samplerate()/fInput;
This is where my understanding is lacking. My assumptions are as follows:
The freq which is passed in corresponds to the fundamental of comb filter response i.e. the first peak. To achieve this I simply divide the SR by this freq to get the first subsequent multiples of the position of the poles. Am I looking at this the wrong way?
best regards, Conor