--- /dev/null
+
+declare name "subtractive -- saw wave filtered with resonant lowpass";
+declare author "Albert Graef";
+declare version "1.0";
+
+import("music.lib");
+
+// control variables
+
+// master volume and pan
+vol = hslider("vol", 0.3, 0, 10, 0.01); // %
+pan = hslider("pan", 0.5, 0, 1, 0.01); // %
+
+// ADSR envelop
+attack = hslider("attack", 0.01, 0, 1, 0.001); // sec
+decay = hslider("decay", 0.3, 0, 1, 0.001); // sec
+sustain = hslider("sustain", 0.5, 0, 1, 0.01); // %
+release = hslider("release", 0.2, 0, 1, 0.001); // sec
+
+// filter parameters
+res = hslider("resonance (dB)", 3, 0, 20, 0.1);
+cutoff = hslider("cutoff (harmonic)", 6, 1, 20, 0.1);
+
+// voice parameters
+freq = nentry("freq", 440, 20, 20000, 1); // Hz
+gain = nentry("gain", 1, 0, 10, 0.01); // %
+gate = button("gate"); // 0/1
+
+// generic table-driven oscillator with phase modulation
+
+// n = the size of the table, must be a power of 2
+// f = the wave function, must be defined on the range [0,2*PI]
+// freq = the desired frequency in Hz
+// mod = the phase modulation signal, in radians
+
+tblosc(n,f,freq,mod) = (1-d)*rdtable(n,waveform,i&(n-1)) +
+ d*rdtable(n,waveform,(i+1)&(n-1))
+with {
+ waveform = time*(2.0*PI)/n : f;
+ phase = freq/SR : (+ : decimal) ~ _;
+ modphase = decimal(phase+mod/(2*PI))*n;
+ i = int(floor(modphase));
+ d = decimal(modphase);
+};
+
+// resonant lowpass
+
+// This is a tweaked Butterworth filter by David Werner and Patrice Tarrabia,
+// see http://www.musicdsp.org and http://www.experimentalscene.com for
+// details.
+
+// res = resonance in dB above DC gain
+// freq = cutoff frequency
+
+lowpass(res,freq) = f : (+ ~ g) : *(a)
+with {
+ f(x) = a0*x+a1*x'+a2*x'';
+ g(y) = 0-b1*y-b2*y';
+ a = 1/db2linear(0.5*res);
+
+ c = 1.0/tan(PI*(freq/SR));
+ c2 = c*c;
+ r = 1/db2linear(2.0*res);
+ q = sqrt(2.0)*r;
+ a0 = 1.0/(1.0+(q*c)+(c2));
+ a1 = 2.0*a0;
+ a2 = a0;
+ b1 = 2.0*a0*(1.0-c2);
+ b2 = a0*(1.0-q*c+c2);
+};
+
+// subtractive synth (saw wave passed through resonant lowpass)
+
+saw(x) = x/PI-1;
+
+smooth(c) = *(1-c) : +~*(c);
+
+process = tblosc(1<<16, saw, freq, 0) : ((env,freq,_) : filter) :
+ *(env * (gain/*:smooth(0.999)*/))
+ : vgroup("3-master", *(vol) : panner(pan))
+with {
+ env = gate : vgroup("1-adsr", adsr(attack, decay, sustain, release));
+ filter(env,freq)
+ = vgroup("2-filter", lowpass(env*res, fmax(1/cutoff, env)*freq*cutoff));
+};