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 State variableType : 12db resonant low, high or bandpassReferences : Effect Deisgn Part 1, Jon Dattorro, J. Audio Eng. Soc., Vol 45, No. 9, 1997 SeptemberNotes : Digital approximation of Chamberlin two-pole low pass. Easy to calculate coefficients, easy to process algorithm.Code : cutoff = cutoff freq in Hz fs = sampling frequency //(e.g. 44100Hz) f = 2 sin (pi * cutoff / fs) //[approximately] q = resonance/bandwidth [0 < q <= 1] most res: q=1, less: q=0 low = lowpass output high = highpass output band = bandpass output notch = notch output scale = q low=high=band=0; //--beginloop low = low + f * band; high = scale * input - low - q*band; band = f * high + band; notch = high + low; //--endloop

 CommentsAdded on : 11/01/06 by nopeComment : Wow, great. Sounds good, thanks.Added on : 13/02/07 by no[ DOT ]spam[ AT ]plea[ DOT ]seComment : The variable "high" doesn't have to be initialised, does it? It looks to me like the only variables that need to be kept around between iterations are "low" and "band".Added on : 13/02/07 by nobody[ AT ]nowhere[ DOT ]comComment : Right. High and notch are calculated from low and band every iteration.Added on : 18/07/07 by lala[ AT ]no[ DOT ]goComment : Anyone know what the difference is between q and scale?Added on : 29/07/07 by jabberdabber[ AT ]hotmail[ DOT ]comComment : "most res: q=1, less: q=0" Someone correct me if I'm wrong, but isn't that backwards? q=0 is max res, q=1 is min res. q and scale are the same value. What the algorithm is doing is scaling the input the higher the resonance is turned up to prevent clipping. One reason why I think 0 equals max resonance and 1 equals no resonance. So as q approaches zero, the input is attenuated more and more. In other words, as you turn up the resonance, the input is turned down. Added on : 16/11/07 by does[ AT ]not[ DOT ]matterComment : scale = sqrt(q); and //value (0;100) - for example q = sqrt(1.0 - atan(sqrt(value)) * 2.0 / PI); f = frqHz / sampleRate*4.; uffffffff :) Now enjoy!Added on : 29/11/08 by kb[ AT ]kebby[ DOT ]orgComment : One drawback of this is that the cutoff frequency can only go up to SR/4 instead of SR/2 - but you can easily compensate it by using 2x oversampling, eg. simply running this thing twice per sample (apply input interpolation or further output filtering ad lib, but from my experience simple linear interpolation of the input values (in and (in+lastin)/2) works well enough).Added on : 05/03/09 by neolit123[ AT ]gmail[ DOT ]comComment : here is the filter with 2x oversampling + some x,y pad functionality to morph between states: like this fx (uses different filter) http://img299.imageshack.us/img299/4690/statevarible.png smoothing with interpolation is suggest for most parameters: //sr: samplerate; //cutoff: 20 - 20k; //qvalue: 0 - 100; //x, y: 0 - 1 q = sqrt(1 - atan(sqrt(qvalue)) * 2 / pi); scale = sqrt(q); f = slider1 / sr * 2; // * 2 here instead of 4 //----------sample loop //set 'input' here //os x2 for (i=0; i<2; i++) { low = low + f * band; high = scale * input - low - q * band; band = f * high + band; notch = high + low; ); //  x,y pad scheme //     //  high -- notch //  |           | //  |           | //  low ---- band // // // use two pairs //low, high pair1 = low * y + high * (1-y); //band, notch pair2 = band * y + notch * (1-y); //out out = pair2 * x + pair1 * (1-x); //----------sample loop

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