Rename interpretor to interpreter.

[Faustine.git] / interpreter / preprocessor / faust-0.9.47mr3 / examples / faust-stk / harpsi.dsp

diff --git a/interpreter/preprocessor/faust-0.9.47mr3/examples/faust-stk/harpsi.dsp b/interpreter/preprocessor/faust-0.9.47mr3/examples/faust-stk/harpsi.dsp
new file mode 100644 (file)
index 0000000..1b5b606
--- /dev/null
+++ b/interpreter/preprocessor/faust-0.9.47mr3/examples/faust-stk/harpsi.dsp
@@ -0,0 +1,90 @@
+declare name "Harpsichord";
+declare description "Nonlinear WaveGuide Commuted Harpsichord";
+declare author "Romain Michon (rmichon@ccrma.stanford.edu)";
+declare copyright "Romain Michon";
+declare version "1.0";
+declare licence "STK-4.3"; // Synthesis Tool Kit 4.3 (MIT style license);
+declare description "A commuted WaveGuide Harpsichord."; 
+
+import("music.lib");
+import("instrument.lib");
+
+//==================== GUI SPECIFICATION ================
+
+freq = nentry("h:Basic_Parameters/freq [1][unit:Hz] [tooltip:Tone frequency]",440,20,20000,1);
+gain = nentry("h:Basic_Parameters/gain [1][tooltip:Gain (value between 0 and 1)]",0.8,0,1,0.01); 
+gate = button("h:Basic_Parameters/gate [1][tooltip:noteOn = 1, noteOff = 0]");
+
+typeModulation = nentry("h:Physical_and_Nonlinearity/v:Nonlinear_Filter_Parameters/Modulation_Type 
+[2][tooltip: 0=theta is modulated by the incoming signal; 1=theta is modulated by the averaged incoming signal;
+2=theta is modulated by the squared incoming signal; 3=theta is modulated by a sine wave of frequency freqMod;
+4=theta is modulated by a sine wave of frequency freq;]",0,0,4,1);
+nonLinearity = hslider("h:Physical_and_Nonlinearity/v:Nonlinear_Filter_Parameters/Nonlinearity 
+[2][tooltip:Nonlinearity factor (value between 0 and 1)]",0,0,1,0.01);
+frequencyMod = hslider("h:Physical_and_Nonlinearity/v:Nonlinear_Filter_Parameters/Modulation_Frequency 
+[2][unit:Hz][tooltip:Frequency of the sine wave for the modulation of theta (works if Modulation Type=3)]",220,20,1000,0.1);
+
+//==================== PROCESSING ================
+
+//----------------------- Nonlinear filter ----------------------------
+//nonlinearities are created by the nonlinear passive allpass ladder filter declared in filter.lib
+
+//nonlinear filter order
+nlfOrder = 6; 
+
+//nonLinearModultor is declared in instrument.lib, it adapts allpassnn from filter.lib 
+//for using it with waveguide instruments
+NLFM =  nonLinearModulator((nonLinearity : smooth(0.999)),1,freq,
+     typeModulation,(frequencyMod : smooth(0.999)),nlfOrder);
+
+//----------------------- Synthesis parameters computing and functions declaration ----------------------------
+
+//convert a frequency in a midi note number
+freqToNoteNumber = (log-log(440))/log(2)*12+69+0.5 : int;
+freqn = freq : freqToNoteNumber;
+
+//string excitation
+soundBoard = dryTapAmp*noise
+       with{
+               dryTapAmpT60 = ffunction(float getValueDryTapAmpT60(float), <harpsichord.h>,"");
+               noteCutOffTime = freqn : dryTapAmpT60*gain;
+               dryTapAmp = asympT60(0.15,0,noteCutOffTime,gate);
+       };      
+
+//loopfilter is a biquad filter whose coefficients are extracted from a C++ file using the foreign function mechanism
+loopFilter = TF2(b0,b1,b2,a1,a2)
+          with{
+               //functions are imported from the C++ file
+               loopFilterb0 = ffunction(float getValueLoopFilterb0(float), <harpsichord.h>,"");
+               loopFilterb1 = ffunction(float getValueLoopFilterb1(float), <harpsichord.h>,"");
+               loopFilterb2 = ffunction(float getValueLoopFilterb2(float), <harpsichord.h>,"");
+               loopFiltera1 = ffunction(float getValueLoopFiltera1(float), <harpsichord.h>,"");
+               loopFiltera2 = ffunction(float getValueLoopFiltera2(float), <harpsichord.h>,"");
+               //coefficients are extracted from the functions
+               b0 = loopFilterb0(freqn);
+               b1 = loopFilterb1(freqn);
+               b2 = loopFilterb2(freqn);
+               a1 = loopFiltera1(freqn);
+               a2 = loopFiltera2(freqn);
+          };
+
+//delay length as a number of samples
+delayLength = SR/freq;
+
+//stereoizer is declared in instrument.lib and implement a stereo spacialisation in function of 
+//the frequency period in number of samples 
+stereo = stereoizer(delayLength);
+
+//----------------------- Algorithm implementation ----------------------------
+
+//envelope for string loop resonance time
+stringLoopGainT = gate*0.9996 + (gate<1)*releaseLoopGain(freqn)*0.9 : smooth(0.999)
+               with{
+                       releaseLoopGain = ffunction(float getValueReleaseLoopGain(float), <harpsichord.h>,"");
+               };
+
+//one string
+string = (*(stringLoopGainT)+_ : delay(4096,delayLength) : loopFilter)~NLFM;
+
+process = soundBoard : string : stereo : instrReverb;
+