Rename interpretor to interpreter.

[Faustine.git] / interpreter / preprocessor / faust-0.9.47mr3 / tools / faust2pd / examples / synth / subtractive.syn

diff --git a/interpreter/preprocessor/faust-0.9.47mr3/tools/faust2pd/examples/synth/subtractive.syn b/interpreter/preprocessor/faust-0.9.47mr3/tools/faust2pd/examples/synth/subtractive.syn
new file mode 100644 (file)
index 0000000..e185609
--- /dev/null
+++ b/interpreter/preprocessor/faust-0.9.47mr3/tools/faust2pd/examples/synth/subtractive.syn
@@ -0,0 +1,85 @@
+
+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));
+};