New directory tree, with preprocessor/ inside interpretor/.

[Faustine.git] / interpretor / faust-0.9.47mr3 / tools / faust2pd / examples / seqdemo / subtractive.dsp

diff --git a/interpretor/faust-0.9.47mr3/tools/faust2pd/examples/seqdemo/subtractive.dsp b/interpretor/faust-0.9.47mr3/tools/faust2pd/examples/seqdemo/subtractive.dsp
deleted file mode 100644 (file)
index e185609..0000000
--- a/interpretor/faust-0.9.47mr3/tools/faust2pd/examples/seqdemo/subtractive.dsp
+++ /dev/null
@@ -1,85 +0,0 @@
-
-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));
-};