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Version 4.0
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Gary Scavone
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Stephen Sinclair
Stephen Sinclair
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commit
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181
src/Drummer.cpp
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181
src/Drummer.cpp
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/***************************************************/
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/*! \class Drummer
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\brief STK drum sample player class.
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This class implements a drum sampling
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synthesizer using WvIn objects and one-pole
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filters. The drum rawwave files are sampled
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at 22050 Hz, but will be appropriately
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interpolated for other sample rates. You can
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specify the maximum polyphony (maximum number
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of simultaneous voices) via a #define in the
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Drummer.h.
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by Perry R. Cook and Gary P. Scavone, 1995 - 2002.
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*/
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/***************************************************/
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#include "Drummer.h"
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#include <string.h>
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#include <math.h>
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// Not really General MIDI yet. Coming soon.
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unsigned char genMIDIMap[128] =
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{ 0,0,0,0,0,0,0,0, // 0-7
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0,0,0,0,0,0,0,0, // 8-15
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0,0,0,0,0,0,0,0, // 16-23
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0,0,0,0,0,0,0,0, // 24-31
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0,0,0,0,1,0,2,0, // 32-39
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2,3,6,3,6,4,7,4, // 40-47
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5,8,5,0,0,0,10,0, // 48-55
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9,0,0,0,0,0,0,0, // 56-63
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0,0,0,0,0,0,0,0, // 64-71
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0,0,0,0,0,0,0,0, // 72-79
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0,0,0,0,0,0,0,0, // 80-87
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0,0,0,0,0,0,0,0, // 88-95
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0,0,0,0,0,0,0,0, // 96-103
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0,0,0,0,0,0,0,0, // 104-111
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0,0,0,0,0,0,0,0, // 112-119
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0,0,0,0,0,0,0,0 // 120-127
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};
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char waveNames[DRUM_NUMWAVES][16] =
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{
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"dope.raw",
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"bassdrum.raw",
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"snardrum.raw",
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"tomlowdr.raw",
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"tommiddr.raw",
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"tomhidrm.raw",
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"hihatcym.raw",
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"ridecymb.raw",
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"crashcym.raw",
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"cowbell1.raw",
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"tambourn.raw"
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};
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Drummer :: Drummer() : Instrmnt()
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{
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for (int i=0; i<DRUM_POLYPHONY; i++) {
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filters[i] = new OnePole;
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sounding[i] = -1;
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}
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// This counts the number of sounding voices.
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nSounding = 0;
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}
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Drummer :: ~Drummer()
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{
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int i;
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for ( i=0; i<nSounding-1; i++ ) delete waves[i];
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for ( i=0; i<DRUM_POLYPHONY; i++ ) delete filters[i];
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}
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void Drummer :: noteOn(MY_FLOAT instrument, MY_FLOAT amplitude)
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{
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#if defined(_STK_DEBUG_)
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cerr << "Drummer: NoteOn instrument = " << instrument << ", amplitude = " << amplitude << endl;
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#endif
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MY_FLOAT gain = amplitude;
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if ( amplitude > 1.0 ) {
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cerr << "Drummer: noteOn amplitude parameter is greater than 1.0!" << endl;
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gain = 1.0;
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}
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else if ( amplitude < 0.0 ) {
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cerr << "Drummer: noteOn amplitude parameter is less than 0.0!" << endl;
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return;
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}
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// Yes, this is tres kludgey.
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int noteNum = (int) ((12*log(instrument/220.0)/log(2.0)) + 57.01);
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// Check first to see if there's already one like this sounding.
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int i, waveIndex = -1;
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for (i=0; i<DRUM_POLYPHONY; i++) {
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if (sounding[i] == noteNum) waveIndex = i;
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}
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if ( waveIndex >= 0 ) {
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// Reset this sound.
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waves[waveIndex]->reset();
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filters[waveIndex]->setPole((MY_FLOAT) 0.999 - (gain * 0.6));
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filters[waveIndex]->setGain(gain);
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}
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else {
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if (nSounding == DRUM_POLYPHONY) {
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// If we're already at maximum polyphony, then preempt the oldest voice.
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delete waves[0];
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filters[0]->clear();
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WvIn *tempWv = waves[0];
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OnePole *tempFilt = filters[0];
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// Re-order the list.
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for (i=0; i<DRUM_POLYPHONY-1; i++) {
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waves[i] = waves[i+1];
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filters[i] = filters[i+1];
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}
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waves[DRUM_POLYPHONY-1] = tempWv;
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filters[DRUM_POLYPHONY-1] = tempFilt;
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}
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else
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nSounding += 1;
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sounding[nSounding-1] = noteNum;
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// Concatenate the STK RAWWAVE_PATH to the rawwave file
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char path[128];
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strcpy(path, RAWWAVE_PATH);
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strcat(path, "rawwaves/");
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strcat(path, waveNames[genMIDIMap[noteNum]]);
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waves[nSounding-1] = new WvIn(path, TRUE);
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if (Stk::sampleRate() != 22050.0)
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waves[nSounding-1]->setRate( 22050.0 / Stk::sampleRate() );
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filters[nSounding-1]->setPole((MY_FLOAT) 0.999 - (gain * 0.6) );
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filters[nSounding-1]->setGain( gain );
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}
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#if defined(_STK_DEBUG_)
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cerr << "Number Sounding = " << nSounding << endl;
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for (i=0; i<nSounding; i++) cerr << sounding[i] << " ";
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cerr << "\n";
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#endif
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}
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void Drummer :: noteOff(MY_FLOAT amplitude)
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{
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// Set all sounding wave filter gains low.
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int i = 0;
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while(i < nSounding) {
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filters[i++]->setGain( amplitude * 0.01 );
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}
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}
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MY_FLOAT Drummer :: tick()
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{
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MY_FLOAT output = 0.0;
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OnePole *tempFilt;
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int j, i = 0;
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while (i < nSounding) {
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if ( waves[i]->isFinished() ) {
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delete waves[i];
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tempFilt = filters[i];
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// Re-order the list.
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for (j=i; j<nSounding-1; j++) {
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sounding[j] = sounding[j+1];
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waves[j] = waves[j+1];
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filters[j] = filters[j+1];
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}
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filters[j] = tempFilt;
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filters[j]->clear();
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sounding[j] = -1;
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nSounding -= 1;
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i -= 1;
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}
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else
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output += filters[i]->tick( waves[i]->tick() );
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i++;
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}
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return output;
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}
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