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BodyseqSynth.cpp
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BodyseqSynth.cpp
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/*
Synth.cpp - Friction Music library
Copyright (c) 2013 Science Friction.
All right reserved.
This library is free software: you can redistribute it and/or modify
it under the terms of the GNU Lesser Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your optionosc1modShape_ptr) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU Lesser Public License for more details.
You should have received a copy of the GNU Lesser Public License
along with Foobar. If not, see <http://www.gnu.org/licenses/>.
+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
+ author: Jakob Bak
+ contact: [email protected]
*/
#include "BodyseqSynth.h"
IntervalTimer synthTimer;
unsigned long timerCounter1 = 0;
unsigned long timerCounter2 = 0;
//MCP4251 Mcp4251 = MCP4251( MCP4251_CS, 100000.0 );
MMusic Music;
MMidi Midi;
const uint16_t sineTable[] = {
#include <FrictionSineTable16bitHex.inc>
};
const uint16_t waveTable[] = {
#include <FrictionWaveTable.inc>
};
// Table of MIDI note values to frequency in Hertz
const float hertzTable[] = {
#include <FrictionHertzTable.inc>
};
uint8_t sequencer[128];
uint8_t instrument[128];
uint8_t userPresets[MAX_PRESETS][PRESET_SIZE];
bool commandFlags[128];
const uint8_t programPresets[] = {
#include <Presets.h>
};
int64_t filterSamplesLP24dB[4];
int64_t filterSamplesHP24dB[8];
int64_t filterSamplesMoogLadder[4];
const int64_t filterCoefficient[] = {
#include <filterCoefficients_1poleLP.inc>
};
const float fcMoog[] = {
#include <filterCutoffFrequenciesMoogLadder.inc>
};
float filterCoefficientsMoogLadderFloat[8][256];
// T = 1 / samplerate
// [0] wd = 2 * PI() * fc
// [1] wa = (2/T) * tan(wd*T/2)
// [2] g = wa * (T/2)
// [3] gg = g * g
// [4] ggg = g * g * g
// [5] G = g * g * g * g
// [6] Gstage = g / (1.0 + g)
// [7] nothing yet
const int64_t filterCoefficientsMoogLadder[] = {
#include <filterCoefficientsMoogLadder.inc>
};
// Used in the functions that set the envelope timing
const uint32_t envTimeTable[] = {1,5,9,14,19,26,34,42,53,65,79,95,113,134,157,182,211,243,278,317,359,405,456,511,570,633,702,776,854,939,1029,1124,1226,1333,1448,1568,1695,1829,1971,2119,2274,2438,2610,2789,2977,3172,3377,3590,3813,4044,4285,4535,4795,5065,5345,5635,5936,6247,6569,6902,7247,7602,7970,8349,8740,9143,9559,9986,10427,10880,11347,11827,12321,12828,13349,13883,14433,14996,15574,16167,16775,17398,18036,18690,19359,20045,20746,21464,22198,22949,23716,24501,25303,26122,26959,27813,28686,29577,30486,31413,32359,33325,34309,35312,36335,37378,38440,39522,40625,41748,42892,44056,45241,46448,47675,48925,50196,51489,52803,54141,55500,56883,58288,59716,61167,62642,64140,65662};
const float semitoneTable[] = {0.25,0.2648658,0.2806155,0.29730177,0.31498027,0.33370996,0.35355338,0.37457678,0.39685026,0.4204482,0.44544938,0.47193715,0.5,0.5297315,0.561231,0.59460354,0.62996054,0.6674199,0.70710677,0.74915355,0.7937005,0.8408964,0.8908987,0.9438743,1.0,1.0594631,1.122462,1.1892071,1.2599211,1.3348398,1.4142135,1.4983071,1.587401,1.6817929,1.7817974,1.8877486,2.0,2.1189263,2.244924,2.3784142,2.5198421,2.6696796,2.828427,2.9966142,3.174802,3.3635857,3.563595,3.7754972,4.0};
const extern uint32_t portamentoTimeTable[] = {1,5,9,13,17,21,26,30,35,39,44,49,54,59,64,69,74,79,85,90,96,101,107,113,119,125,132,138,144,151,158,165,172,179,187,194,202,210,218,226,234,243,252,261,270,279,289,299,309,320,330,341,353,364,376,388,401,414,427,440,455,469,484,500,516,532,549,566,584,602,622,642,663,684,706,729,753,778,804,831,859,888,919,951,984,1019,1056,1094,1134,1176,1221,1268,1317,1370,1425,1484,1547,1614,1684,1760,1841,1929,2023,2125,2234,2354,2484,2627,2785,2959,3152,3368,3611,3886,4201,4563,4987,5487,6087,6821,7739,8918,10491,12693,15996,21500,32509,65535};
void MMusic::generateFilterCoefficientsMoogLadder() {
for(int i=0; i<256; i++) {
float T = 1.0f/float(SAMPLE_RATE);
float wd = 2.0f * PI * fcMoog[i];
float wa = (2.0f/T) * tan(wd*T/2.0f);
// float g = tan(wd*T/2.0f);
float g = wa * (T/2.0f);
float gg = g * g;
float ggg = g * g * g;
float G = g * g * g * g;
float Gstage = g / (1.0 + g);
filterCoefficientsMoogLadderFloat[0][i] = wd;
filterCoefficientsMoogLadderFloat[1][i] = wa;
filterCoefficientsMoogLadderFloat[2][i] = g;
filterCoefficientsMoogLadderFloat[3][i] = gg;
filterCoefficientsMoogLadderFloat[4][i] = ggg;
filterCoefficientsMoogLadderFloat[5][i] = G;
filterCoefficientsMoogLadderFloat[6][i] = Gstage;
filterCoefficientsMoogLadderFloat[7][i] = 0;
}
}
//////////////////////////////////////////////////////////
//
// SYNTH INTERRUPT
//
//////////////////////////////////////////////////////////
void synth_isr(void) {
// if(timerCounter1 > timerCounter2) {
// Serial.println("timerCounter1 > timerCounter2");
// timerCounter2 = timerCounter1;
// }
// timerCounter1++;
Music.output2T3DAC();
// Music.output2DAC();
Music.envelope1();
Music.envelope2();
if(Music.is12bit) Music.synthInterrupt12bitSineFM();
// if(Music.is12bit) Music.phaseDistortionOscillator();
// if(Music.is12bit) Music.synthInterrupt12bitSawFM();
else Music.synthInterrupt8bitFM();
Music.amplifier();
if(Music.lowpass) Music.filterLP6dB();
if(Music.highpass) Music.filterHP6dB();
if(Music.lowpass24dB) Music.filterLP24dB();
if(Music.highpass24dB) Music.filterHP24dB();
if(Music.moogLadder) Music.filterMoogLadder();
// timerCounter2++;
}
void MMusic::set12bit(bool b) {
is12bit = b;
}
/////////////////////////////////////////////////////////
//
// 8 BIT OSCILLATOR - WAVETABLE - 16 WAVEFORMS
//
/////////////////////////////////////////////////////////
void MMusic::synthInterrupt8bitFM ()
{
dPhase1 = dPhase1 + (period1 - dPhase1) / portamento;
modulator1 = (fmAmount1 * fmOctaves1 * (*osc1modSource_ptr))>>10;
modulator1 = (modulator1 * (*osc1modShape_ptr))>>16;
modulator1 = (modulator1 * int64_t(dPhase1))>>16;
modulator1 = (modulator1>>((modulator1>>31)&zeroFM));
accumulator1 = accumulator1 + dPhase1 + modulator1;
index1 = accumulator1 >> 24;
oscil1 = waveTable[index1 + waveForm1];
oscil1 -= 128;
oscil1 <<= 8;
sample = (oscil1 * gain1);
dPhase2 = dPhase2 + (period2 - dPhase2) / portamento;
modulator2 = (fmAmount2 * fmOctaves2 * (*osc2modSource_ptr))>>10;
modulator2 = (modulator2 * (*osc2modShape_ptr))>>16;
modulator2 = (modulator2 * int64_t(dPhase2))>>16;
modulator2 = (modulator2>>((modulator2>>31)&zeroFM));
accumulator2 = accumulator2 + dPhase2 + modulator2;
index2 = accumulator2 >> 24;
oscil2 = waveTable[index2 + waveForm2];
oscil2 -= 128;
oscil2 <<= 8;
sample += (oscil2 * gain2);
dPhase3 = dPhase3 + (period3 - dPhase3) / portamento;
modulator3 = (fmAmount3 * fmOctaves3 * (*osc3modSource_ptr))>>10;
modulator3 = (modulator3 * (*osc3modShape_ptr))>>16;
modulator3 = (modulator3 * int64_t(dPhase3))>>16;
modulator3 = (modulator3>>((modulator3>>31)&zeroFM));
accumulator3 = accumulator3 + dPhase3 + modulator3;
index3 = accumulator3 >> 24;
oscil3 = waveTable[index3 + waveForm3];
oscil3 -= 128;
oscil3 <<= 8;
sample += (oscil3 * gain3);
sample >>= 18;
}
/////////////////////////////////////////////////////////
//
// 12 BIT OSCILLATOR - SINETABLE
//
/////////////////////////////////////////////////////////
void MMusic::synthInterrupt12bitSineFM()
{
dPhase1 = dPhase1 + (period1 - dPhase1) / portamento;
modulator1 = (fmAmount1 * fmOctaves1 * (*osc1modSource_ptr))>>10;
modulator1 = (modulator1 * (*osc1modShape_ptr))>>16;
modulator1 = (modulator1 * int64_t(dPhase1))>>16;
modulator1 = (modulator1>>((modulator1>>31)&zeroFM));
accumulator1 = accumulator1 + dPhase1 + modulator1;
index1 = accumulator1 >> 20;
oscil1 = sineTable[index1];
index1 = accumulator1 >> 20;
oscil1 -= 32768;
sample = (oscil1 * gain1);
dPhase2 = dPhase2 + (period2 - dPhase2) / portamento;
modulator2 = (fmAmount2 * fmOctaves2 * (*osc2modSource_ptr))>>10;
modulator2 = (modulator2 * (*osc2modShape_ptr))>>16;
modulator2 = (modulator2 * int64_t(dPhase2))>>16;
modulator2 = (modulator2>>((modulator2>>31)&zeroFM));
accumulator2 = accumulator2 + dPhase2+ modulator2;
index2 = accumulator2 >> 20;
oscil2 = sineTable[index2];
oscil2 -= 32768;
sample += (oscil2 * gain2);
dPhase3 = dPhase3 + (period3 - dPhase3) / portamento;
modulator3 = (fmAmount3 * fmOctaves3 * (*osc3modSource_ptr))>>10;
modulator3 = (modulator3 * (*osc3modShape_ptr))>>16;
modulator3 = (modulator3 * int64_t(dPhase3))>>16;
modulator3 = (modulator3>>((modulator3>>31)&zeroFM));
accumulator3 = accumulator3 + dPhase3 + modulator3;
index3 = accumulator3 >> 20;
oscil3 = sineTable[index3];
oscil3 -= 32768;
sample += (oscil3 * gain3);
sample >>= 18;
}
void MMusic::synthInterrupt12bitSawFM()
{
dPhase1 = dPhase1 + (period1 - dPhase1) / portamento;
modulator1 = (fmAmount1 * fmOctaves1 * (*osc1modSource_ptr))>>10;
modulator1 = (modulator1 * (*osc1modShape_ptr))>>16;
modulator1 = (modulator1 * int64_t(dPhase1))>>16;
modulator1 = (modulator1>>((modulator1>>31)&zeroFM));
accumulator1 = accumulator1 + dPhase1 + modulator1;
// index1 = accumulator1 >> 20;
// oscil1 = sineTable[index1];
oscil1 = accumulator1 >> 16;
oscil1 -= 32768;
sample = (oscil1 * gain1);
dPhase2 = dPhase2 + (period2 - dPhase2) / portamento;
modulator2 = (fmAmount2 * fmOctaves2 * (*osc2modSource_ptr))>>10;
modulator2 = (modulator2 * (*osc2modShape_ptr))>>16;
modulator2 = (modulator2 * int64_t(dPhase2))>>16;
modulator2 = (modulator2>>((modulator2>>31)&zeroFM));
accumulator2 = accumulator2 + dPhase2+ modulator2;
// index2 = accumulator2 >> 20;
// oscil2 = sineTable[index2];
oscil2 = accumulator2 >> 16;
oscil2 -= 32768;
sample += (oscil2 * gain2);
dPhase3 = dPhase3 + (period3 - dPhase3) / portamento;
modulator3 = (fmAmount3 * fmOctaves3 * (*osc3modSource_ptr))>>10;
modulator3 = (modulator3 * (*osc3modShape_ptr))>>16;
modulator3 = (modulator3 * int64_t(dPhase3))>>16;
modulator3 = (modulator3>>((modulator3>>31)&zeroFM));
accumulator3 = accumulator3 + dPhase3 + modulator3;
// index3 = accumulator3 >> 20;
// oscil3 = sineTable[index3];
oscil3 = accumulator3 >> 16;
oscil3 -= 32768;
sample += (oscil3 * gain3);
sample >>= 18;
}
/////////////////////////////////////////////////////////
//
// ENVELOPES
//
/////////////////////////////////////////////////////////
void MMusic::envelope1() {
if(envelopeOn1) {
// Attack
if(env1Stage == 1) {
env1 += 1; // to make sure the envelope increases when (MAX_ENV_GAIN-env1) is smaller than attack1
env1 += (MAX_ENV_GAIN - env1)/attack1;
if(velPeak1 < env1) {
env1 = velPeak1;
env1Stage = 2;
}
}
// Decay
else if(env1Stage == 2) {
env1 += -1; // to make sure the envelope decreases when (velSustain1-env1) is smaller than decay1
env1 += (velSustain1-env1)/decay1;
if(env1 < velSustain1 || MAX_ENV_GAIN < env1) {
env1 = velSustain1;
env1Stage = 3;
}
}
// Sustain
else if (env1Stage == 3) {
env1 = velSustain1;
}
// Release
else if (env1Stage == 4) {
env1 += -1; // to make sure the envelope decreases when (0-env1) is smaller than release1
env1 += (0 - env1) / release1;
if(env1 < 0 || MAX_ENV_GAIN < env1) {
env1 = 0;
env1Stage = 0;
}
}
// No gain
else if (env1Stage == 0) {
env1 = 0;
}
} else {
env1 = 65535;
}
}
void MMusic::envelope2() {
if(envelopeOn2) {
// Attack
if(env2Stage == 1) {
env2 += 1; // to make sure the envelope increases when (MAX_ENV_GAIN-env2) is smaller than attack1
env2 += (MAX_ENV_GAIN-env2)/attack2;
if(velPeak2 < env2) {
env2 = velPeak2;
env2Stage = 2;
}
}
// Decay
else if(env2Stage == 2) {
env2 += -1; // to make sure the envelope decreases when (velSustain2-env2) is smaller than decay2
env2 += (velSustain2-env2)/decay2;
if(env2 < velSustain2 || MAX_ENV_GAIN < env2) {
env2 = velSustain2;
env2Stage = 3;
}
}
// Sustain
else if (env2Stage == 3) {
env2 = velSustain2;
}
// Release
else if (env2Stage == 4) {
env2 += -1; // to make sure the envelope decreases when (0-env2) is smaller than release2
env2 += (0 - env2) / release2;
if(env2 < 0 || MAX_ENV_GAIN < env2) {
env2 = 0;
env2Stage = 0;
}
}
// No gain
else if (env2Stage == 0) {
env2 = 0;
//accumulator1 = 0;
//accumulator2 = 0;
//accumulator3 = 0;
}
} else {
env2 = 65535;
}
}
void MMusic::amplifier() {
sample = (env1 * sample) >> 16;
}
/////////////////////////////////////////////////////////
//
// SEND SAMPLE TO DAC ON TEENSY 3.1 PIN A14
//
/////////////////////////////////////////////////////////
void MMusic::output2T3DAC() {
sample += 32768;
analogWrite(A14, sample>>4);
}
void MMusic::output2DAC() {
sample += 32768;
dacSPIA0 = sample >> 8;
dacSPIA0 >>= 4;
dacSPIA0 |= dacSetA;
dacSPIA1 = sample >> 4;
digitalWriteFast(DAC_CS, LOW);
spi4teensy3::send(dacSPIA0);
spi4teensy3::send(dacSPIA1);
// while(SPI.transfer(dacSPIB0));
// while(SPI.transfer(dacSPIB1));
digitalWriteFast(DAC_CS, HIGH);
}
/////////////////////////////////////
//
// INITIALIZING FUNCTION
//
/////////////////////////////////////
void MMusic::spi_setup()
{
spi4teensy3::init(0,0,0);
pinMode(DAC_CS, OUTPUT);
}
void MMusic::getPreset(uint8_t p)
{
const static int AO_CONTROL_MESSAGES = 40;
const static int AO_CONTROL_MESSAGES_MAX = 69;
// cli();
if(p < MAX_PRESETS) {
// Serial.print("GETTING PRESET NUMBER : ");
// Serial.println(p);
for(uint8_t i=2; i<128; i++) {
if(i >= 80 && i < 100) continue;
if (i >= AO_CONTROL_MESSAGES && i < AO_CONTROL_MESSAGES_MAX) continue;
instrument[i] = userPresets[p][i];
Midi.controller(Midi.midiChannel, i, instrument[i]);
// Serial.println(userPresets[p][i]);
}
} else {
for(uint8_t i=2; i<128; i++) {
if(i >= 80 && i < 100) continue;
if (i >= AO_CONTROL_MESSAGES && i < AO_CONTROL_MESSAGES_MAX) continue;
instrument[i] = programPresets[(p-MAX_PRESETS)*PRESET_SIZE + i];
Midi.controller(Midi.midiChannel, i, instrument[i]);
// Serial.println(userPresets[p][i]);
}
}
// sei();
}
void MMusic::getRandomizedPreset(uint8_t p, uint8_t r)
{
Serial.print("randomize preset ");
Serial.print(p);
Serial.print(" by ");
Serial.print(r);
Serial.println("%");
// cli();
// load preset values
if(p < MAX_PRESETS) {
for(uint8_t i=2; i<128; i++) {
instrument[i] = userPresets[p][i];
}
} else {
for(uint8_t i=2; i<128; i++) {
instrument[i]=programPresets[(p-MAX_PRESETS)*PRESET_SIZE+i];
}
}
// apply randomization
for(uint8_t i=2; i<128; i++) {
uint8_t presetValue = instrument[i];
int ran = 0;
// randomization only affects the following settings.
// randomizing all paramaters will make a lot of unusable sounds.
// 4: cutoff, 70: cutoff mod, 6: fm oct, 8: portamento
if (i==4||i==70||i==6||i==8) {
ran = random(float(r)/100*-127,float(r)/100*127);
}
// 10: LFO1, 12: detune1, 13: gain1, 15: FM1
// 20: LFO2, 22: detune2, 23: gain2, 25: FM2
// 30: LFO3, 32: detune3, 33: gain3, 35: FM3
if (i==10||i==12||i==13||i==15||i==20||i==22||i==23||i==25||i==30||i==32||i==33||i==35) {
ran = random(float(r)/100*-127,float(r)/100*127);
}
// 114: attack1, 115: decay1, 116: sustain1, 117: release1
// 124: attack2, 125: decay2, 126: sustain2, 127: release2
if (i==114||i==115||i==116||i==117||i==124||i==125||i==126||i==127) {
ran = random(float(r)/100*-127,float(r)/100*127);
}
uint8_t newVal = constrain(presetValue+ran,0,127);
if (ran != 0)
{
/*
Serial.print(i);
Serial.print(": preset value ");
Serial.print(presetValue);
Serial.print(" randomized by ");
Serial.print(ran);
Serial.print(" yielding ");
Serial.println(newVal);
*/
}
instrument[i] = newVal;
Midi.controller(Midi.midiChannel, i, instrument[i]);
}
// sei();
}
#if defined(USB_MIDI)
void MMusic::sendInstrument()
{
Serial.print("SENDING CURRENT INSTRUMENT ");
Serial.print(" OVER MIDI on channel ");
Serial.println(Midi.midiChannel+1);
cli();
for(uint8_t i=2; i<128; i++) {
usbMIDI.sendControlChange(i, instrument[i], Midi.midiChannel+1);
}
sei();
}
#else
void MMusic::sendInstrument(){;}
#endif
void MMusic::savePreset(uint8_t p)
{
if(p < MAX_PRESETS) {
Serial.print("SAVING PRESET NUMBER : ");
Serial.println(p);
for(uint8_t i=0; i<128; i++) {
//Serial.print(i);
//Serial.print(" : ");
userPresets[p][i] = instrument[i];
//Serial.println(userPresets[p][i]);
//insert code for saving instrument sequence here
cli();
EEPROM.write(p * PRESET_SIZE + i, instrument[i]);
sei();
}
}
else {
Serial.println("CAN NOT SAVE PRESET TO EEPROM - COPY/PASTE BELOW TO FILE");
for(uint8_t i=0; i<128; i++) {
Serial.print(instrument[i]);
Serial.print(", ");
}
}
}
void MMusic::loadAllPresets()
{
for(uint8_t i=2; i<128; i++) {
for(uint8_t p=0; p<MAX_PRESETS;p++) {
userPresets[p][i] = EEPROM.read(p * PRESET_SIZE + i);
}
}
}
void MMusic::init()
{
// pinMode(MUX_A, OUTPUT);
// pinMode(MUX_B, OUTPUT);
Midi.init();
for(uint8_t i=0; i<128; i++) {
instrument[i] = 0;
for(uint8_t p=0; p<MAX_PRESETS;p++) {
userPresets[p][i] = 0;
}
}
generateFilterCoefficientsMoogLadder();
for(int i=0; i<256; i++) {
Serial.println(filterCoefficientsMoogLadderFloat[6][i]);
}
sampleRate = SAMPLE_RATE;
sample = 0;
set12bit(false);
setPortamento(0);
dPhase1 = 0;
dPhase2 = 0;
dPhase3 = 0;
modulator1 = 0;
modulator2 = 0;
modulator3 = 0;
fullSignal = 65535;
invertSignal = -65535;
noSignal = 0;
osc1modSource_ptr = &oscil3;
osc2modSource_ptr = &oscil1;
osc3modSource_ptr = &oscil2;
osc1modShape_ptr = &fullSignal;
osc2modShape_ptr = &fullSignal;
osc3modShape_ptr = &fullSignal;
amp_modSource_ptr = &env1;
amp_modShape_ptr = &fullSignal;
setFM1Source(3);
setFM2Source(1);
setFM3Source(2);
setFM1Shape(0);
setFM2Shape(0);
setFM3Shape(0);
fmToZeroHertz(true);
accumulator1 = 0;
accumulator2 = 0;
accumulator3 = 0;
index1 = 0;
index2 = 0;
index3 = 0;
oscil1 = 0;
oscil2 = 0;
oscil3 = 0;
// waveform setup
setWaveform(0);
// frequency setup
setFrequency(440);
setSemitone1(0);
setSemitone2(0);
setSemitone3(0);
setDetune(0);
setOsc1LFO(false);
setOsc2LFO(false);
setOsc3LFO(false);
// gain setup
setGain(1.0f);
setGain1(1.0f);
setGain2(1.0f);
setGain3(1.0f);
// envelope setup
setEnv1Stage(0);
disableEnvelope1();
env1 = 0;
setEnv1Attack(4);
setEnv1Decay(90);
setEnv1Sustain(32);
setEnv1Release(64);
setEnv1VelSustain(0);
setEnv2Stage(0);
disableEnvelope2();
env2 = 0;
setEnv2Attack(8);
setEnv2Decay(36);
setEnv2Sustain(0);
setEnv2Release(64);
setEnv2VelSustain(0);
//FM setup
setFM1(0);
setFM2(0);
setFM3(0);
setFMoctaves(0);
// DAC setup
dacSetA = 0;
dacSetB = 0;
dacSetA |= (DAC_A << DAC_AB) | (0 << DAC_BUF) | (1 << DAC_GA) | (1 << DAC_SHDN);
dacSetB |= (DAC_B << DAC_AB) | (0 << DAC_BUF) | (1 << DAC_GA) | (1 << DAC_SHDN);
analogWriteResolution(12);
// analogWrite(A14, 0); //Set the DAC output to 0.
//// DAC0_C0 &= 0b10111111; //uses 1.2V reference for DAC instead of 3.3V
//
// SIM_SCGC2 |= SIM_SCGC2_DAC0;
// DAC0_C0 = DAC_C0_DACEN; // 1.2V VDDA is DACREF_2
//// DAC0_C0 = DAC_C0_DACEN | DAC_C0_DACRFS; // 3.3V VDDA is DACREF_2
// // slowly ramp up to DC voltage, approx 1/4 second
// for (int16_t i=0; i<128; i++) {
// analogWrite(A14, i);
// delay(2);
// }
loadAllPresets();
// spi_setup();
// filter setup
setCutoff((uint16_t)BIT_16);
// setCutoff(BIT_16);
setResonance(BIT_16);
setFilterType(0);
cutoffModSource_ptr = &env2;
resonanceModSource_ptr = &fullSignal;
cutoffModShape_ptr = &fullSignal;
resonanceModShape_ptr = &fullSignal;
setCutoffModSource(2);
setCutoffModAmount(BIT_16);
setCutoffModDirection(1);
cli();
synthTimer.begin(synth_isr, 1000000.0 / sampleRate);
sei();
}
/////////////////////////////////////
//
// FILTER FUNCTIONS
//
/////////////////////////////////////
void MMusic::setCutoff(uint16_t c)
{
cutoff = c;
}
void MMusic::setResonance(uint32_t res)
{
resonance = res;
k = res;
}
void MMusic::setCutoffModAmount(int32_t amount) {
if(amount >= 65536) cutoffModAmount = 65535;
else if(amount < -65536) cutoffModAmount = -65536;
else cutoffModAmount = amount;
// cutoffModAmount = amount;
}
void MMusic::setCutoffModDirection(int32_t direction) {
if(direction >= 0) cutoffModDirection = 1;
else cutoffModDirection = -1;
}
void MMusic::filterLP6dB() {
int64_t mod = (int64_t(cutoffModAmount) * (int64_t(*cutoffModSource_ptr)))>>16;
int64_t c = (mod + int64_t(cutoff));
if(c > 65535) c = 65535;
else if(c < 0) c = 0;
// c = ((((c * 32768) >> 15) + 65536) >> 1);
b1 = filterCoefficient[c>>8];
a0 = BIT_32 - b1;
sample = (a0 * sample + b1 * lastSampleOutLP) >> 32;
lastSampleOutLP = sample;
}
void MMusic::filterLP24dB() { // BROKEN?????
int64_t mod = (int64_t(cutoffModAmount) * (int64_t(*cutoffModSource_ptr)))>>16;
int64_t c = (mod + int64_t(cutoff));
if(c > 65535) c = 65535;
else if(c < 0) c = 0;
b1 = filterCoefficient[c>>8];
a0 = BIT_32 - b1;
x0 = sample;
y1 = filterSamplesLP24dB[0];
y2 = filterSamplesLP24dB[1];
y3 = filterSamplesLP24dB[2];
y4 = filterSamplesLP24dB[3];
y1 = (a0 * x0 + b1 * y1) >> 32;
y2 = (a0 * y1 + b1 * y2) >> 32;
y3 = (a0 * y2 + b1 * y3) >> 32;
y4 = (a0 * y3 + b1 * y4) >> 32;
filterSamplesLP24dB[0] = y1;
filterSamplesLP24dB[1] = y2;
filterSamplesLP24dB[2] = y3;
filterSamplesLP24dB[3] = y4;
sample = y4;
}
void MMusic::filterHP24dB() {
int64_t mod = (int64_t(cutoffModAmount) * (int64_t(*cutoffModSource_ptr)))>>16;
int64_t c = (mod + int64_t(cutoff));
if(c > 65535) c = 65535;
else if(c < 0) c = 0;
b1 = filterCoefficient[c>>8];
a0 = (BIT_32 + b1) >> 1;
a1 = -a0;
xNew = sample;
xOld = filterSamplesHP24dB[0];
yOld = filterSamplesHP24dB[4];
yNew = (a0 * xNew + a1 * xOld + b1 * yOld) >> 32;
x1 = xNew;
y1 = yNew;
xNew = y1;
xOld = filterSamplesHP24dB[1];
yOld = filterSamplesHP24dB[5];
yNew = (a0 * xNew + a1 * xOld + b1 * yOld) >> 32;
x2 = xNew;
y2 = yNew;
xNew = y2;
xOld = filterSamplesHP24dB[2];
yOld = filterSamplesHP24dB[6];
yNew = (a0 * xNew + a1 * xOld + b1 * yOld) >> 32;
x3 = xNew;
y3 = yNew;
xNew = y3;
xOld = filterSamplesHP24dB[3];
yOld = filterSamplesHP24dB[7];
yNew = (a0 * xNew + a1 * xOld + b1 * yOld) >> 32;
x4 = xNew;
y4 = yNew;
filterSamplesHP24dB[0] = x1;
filterSamplesHP24dB[1] = x2;
filterSamplesHP24dB[2] = x3;
filterSamplesHP24dB[3] = x4;
filterSamplesHP24dB[4] = y1;
filterSamplesHP24dB[5] = y2;
filterSamplesHP24dB[6] = y3;
filterSamplesHP24dB[7] = y4;
sample = y4;
}
void MMusic::filterMoogLadder() {
int64_t mod = (int64_t(cutoffModAmount) * (int64_t(*cutoffModSource_ptr)))>>16;
int64_t c = (mod + int64_t(cutoff));
if(c > 65535) c = 65535;
else if(c < 0) c = 0;
int fc = c>>8;
if(fc > 234) fc = 234;
x0 = sample;
u = x0;
Gstage = filterCoefficientsMoogLadder[1024 + fc];
v1 = ((u - z1) * Gstage) >> 32;
y1 = (v1 + z1);
z1 = y1 + v1;
v2 = ((y1 - z2) * Gstage) >> 32;
y2 = (v2 + z2);
z2 = y2 + v2;
v3 = ((y2 - z3) * Gstage) >> 32;
y3 = (v3 + z3);
z3 = y3 + v3;
v4 = ((y3 - z4) * Gstage) >> 32;
y4 = (v4 + z4);
z4 = y4 + v4;
sample = y4;
}
void MMusic::filterHP6dB() {
sampleInHP = sample;
int64_t mod = (int64_t(cutoffModAmount) * (int64_t(*cutoffModSource_ptr)))>>16;
int64_t c = (mod + int64_t(cutoff));
if(c > 65535) c = 65535;
else if(c < 0) c = 0;
// c = ((((c * 32768) >> 15) + 65536) >> 1);
b1 = filterCoefficient[c>>8];
a0 = (BIT_32 + b1) >> 1;
a1 = -a0;
sampleOutHP = (a0 * sampleInHP + a1 * lastSampleInHP + b1 * lastSampleOutHP) >> 32;
lastSampleInHP = sampleInHP;
lastSampleOutHP = sampleOutHP;
sample = sampleOutHP;
}
void MMusic::setFilterType(uint8_t type) {
switch (type) {
case LP6:
lowpass = true;
highpass = false;
lowpass24dB = false;
highpass24dB = false;
moogLadder = false;
break;