/************************************************************************ ************************************************************************ FAUST library file Copyright (C) 2003-2011 GRAME, Centre National de Creation Musicale --------------------------------------------------------------------- This program is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2.1 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details. You should have received a copy of the GNU Lesser General Public License along with the GNU C Library; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. ************************************************************************ ************************************************************************/ declare name "MaxMSP compatibility Library"; declare author "GRAME"; declare copyright "GRAME"; declare version "1.0"; declare license "LGPL"; import("music.lib"); atodb = db2lin; //------------------------------------------------------------------------- // // Implementation of MaxMSP filtercoeff // // from : Cookbook formulae for audio EQ biquad filter coefficients // by : Robert Bristow-Johnson // URL : http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt // //------------------------------------------------------------------------- filtercoeff(f0, dBgain, Q) = environment { //---------------------------------------- // biquad coeffs for various filters // usage : filtercoeff(f0, dBgain, Q).LPF //---------------------------------------- LPF = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = (1 - cos(w0))/2; b1 = 1 - cos(w0); b2 = (1 - cos(w0))/2; a0 = 1 + alpha; a1 = -2*cos(w0); a2 = 1 - alpha; }; HPF = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = (1 + cos(w0))/2; b1 = -1 - cos(w0); b2 = (1 + cos(w0))/2; a0 = 1 + alpha; a1 = -2*cos(w0); a2 = 1 - alpha; }; BPF = rbjcoef( a0, a1, a2, b0, b1, b2 ) // constant 0 dB peak gain with { b0 = alpha; b1 = 0; b2 = -alpha; a0 = 1 + alpha; a1 = -2*cos(w0); a2 = 1 - alpha; }; notch = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = 1; b1 = -2*cos(w0); b2 = 1; a0 = 1 + alpha; a1 = -2*cos(w0); a2 = 1 - alpha; }; APF = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = 1 - alpha; b1 = -2*cos(w0); b2 = 1 + alpha; a0 = 1 + alpha; a1 = -2*cos(w0); a2 = 1 - alpha; }; peakingEQ = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = 1 + alpha*A; b1 = -2*cos(w0); b2 = 1 - alpha*A; a0 = 1 + alpha/A; a1 = -2*cos(w0); a2 = 1 - alpha/A; }; peakNotch = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = 1 + alpha*G; b1 = -2*cos(w0); b2 = 1 - alpha*G; a0 = 1 + alpha/G; a1 = -2*cos(w0); a2 = 1 - alpha/G; }; lowShelf = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha ); b1 = 2*A*( (A-1) - (A+1)*cos(w0) ); b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha ); a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha; a1 = -2*( (A-1) + (A+1)*cos(w0) ); a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha; }; highShelf = rbjcoef( a0, a1, a2, b0, b1, b2 ) with { b0 = A*( (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha ); b1 = -2*A*( (A-1) + (A+1)*cos(w0) ); b2 = A*( (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha ); a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha; a1 = 2*( (A-1) - (A+1)*cos(w0) ); a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha; }; // --------------------- implementation ------------------------------ // convert rbj coeffs to biquad coeffs rbjcoef(a0,a1,a2,b0,b1,b2) = (b0/a0, b1/a0, b2/a0,-a1/a0,-a2/a0); // common values // alpha = sin(w0)/(2*Q); // w0 = 2*PI*f0/Fs; alpha = sin(w0)/(2*max(0.001,Q)); w0 = 2*PI*max(0,f0)/Fs; Fs = SR; A = 10^(dBgain/40); // (for peaking and shelving EQ filters only) G = sqrt(max(0.00001, dBgain)); // When gain is a linear values (i.e. not in dB) }; //------------------------------------------------------------------------- // Implementation of MaxMSP biquad~ // y[n] = a0 * x[n] + a1 * x[n-1] + a2 * x[n-2] + b1 * y[n-1] + b2 * y[n-2] //------------------------------------------------------------------------- biquad(x,a0,a1,a2,b1,b2) = x : conv3(a0, a1, a2) : + ~ conv2(b1, b2) with { conv2(c0,c1,x) = c0*x+c1*x'; conv3(c0,c1,c2,x) = c0*x+c1*x'+c2*x''; }; //------------------------------------------------------------------------- // // Filters using filtercoeff and biquad // //------------------------------------------------------------------------- // Low Pass Filter LPF(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).LPF : biquad; // High Pass Filter HPF(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).HPF : biquad; // Band Pass Filter BPF(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).BPF : biquad; // notch Filter notch(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).notch : biquad; // All Pass Filter APF(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).APF : biquad; // ???? peakingEQ(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).peakingEQ : biquad; // Max peakNotch is like peakingEQ but with a linear gain peakNotch(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).peakNotch : biquad; // ???? lowShelf(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).lowShelf : biquad; // ???? highShelf(x, f0, gain, Q) = x , filtercoeff(f0,gain,Q).highShelf : biquad; //------------------------------------------------------------------------- // Implementation of Max/MSP line~. Generate signal ramp or envelope // // USAGE : line(value, time) // value : the desired output value // time : the interpolation time to reach this value (in milliseconds) // // NOTE : the interpolation process is restarted every time the desired // output value changes. The interpolation time is sampled only then. //------------------------------------------------------------------------- line (value, time) = state ~ ( _ , _ ) : ! , _ with { state (t , c) = nt , if( nt <= 0 , value , c + (value - c) / nt) with { nt = if( value != value' , samples, t - 1) ; samples = time * SR / 1000.0 ; } ; } ;