Fixed Modal example, added Polysynth example

Fixed voice allocation code in Modal example. Added polysynth example.
main
Rich Heslip 4 years ago
parent 6aa5a65294
commit 4ba885761f
  1. 7
      examples/modal/modal.ino
  2. 295
      examples/polysynth/polysynth.ino

@ -31,7 +31,7 @@ float samplerate=AUDIO_SAMPLE_RATE_EXACT;
// create daisySP processing objects
#define VOICES 1 // uses too much CPU for more than 2 and more than 1 doesn't work anyway
#define VOICES 3 // 87% CPU with 3 voices 811mhz overclock
daisysp::ModalVoice voice[VOICES];
ReverbSc verb;
@ -167,12 +167,15 @@ void loop() {
// Handling the voice allocation here.
void myNoteOn(byte channel, byte note, byte velocity) {
for (int i=0; i <= VOICES; ++i){
int i=0;
while( i < VOICES){
if (StoredNotes[i] == -1) { // if voice is idle
StoredNotes[i] = int(note); // allocate this voice
voice[i].SetFreq(NoteNumToFreq[note]);
voice[i].Trig();
break;
}
++i;
}
}

@ -0,0 +1,295 @@
// test of DaisySP synth object for the Teensy audio library
// simple poly synth
// some of this code was cribbed from the Faust for Teensy Additivesynth example
// RH March 29 2021
#include <Audio.h>
#include <Metro.h>
//#define DEBUG // comment out to remove debug code
#ifdef DEBUG
Metro five_sec=Metro(5000); // Set up a 5 second Metro for performance stats
#endif
// constants for integer to float and float to integer conversion
#define MULT_16 2147483647
#define DIV_16 4.6566129e-10
// for polyphony - an array of all current notes.
// Value -1 means the note is off (not sounding).
#define VOICES 16 //
int StoredNotes[VOICES];
#include "daisysp.h"
using namespace daisysp;
// including the source files is a pain but that way you compile in only the modules you need
// DaisySP statically allocates memory and some modules e.g. reverb use a lot of ram
#include "synthesis/oscillator.cpp"
#include "control/adsr.cpp"
#include "filters/moogladder.cpp"
#include "effects/reverbsc.cpp" // uses a LOT of ram
float samplerate=AUDIO_SAMPLE_RATE_EXACT;
// parameters we can modify via MIDI CCs
int waveform=0;
float detune=0;
float filterfreq=100;
float filtersweep=3000;
float filterresonance=0.3;
float reverblevel=0.1;
float lfofreq=0.1;
float lfofreqdepth=0;
float lfofilterdepth=0;
// create daisySP processing objects
#define OSCSPERVOICE 3 // note - the detune code is set up for 3 oscillators
Oscillator osc[VOICES * OSCSPERVOICE];
Oscillator lfo;
Adsr env[VOICES];
MoogLadder filt[VOICES];
ReverbSc verb;
// this is the function called by the AudioSynthDaisySP object when it needs a block of samples
void AudioSynthDaisySP::update(void)
{
float out,sig,outsig,envelope,filtsig,wetvl, wetvr;
bool gate;
audio_block_t *block;
block = allocate(); // grab an audio block
if (!block) {
return;
}
for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
//**** insert daisySP generators here
outsig=0; // sum up voices
for (int i=0; i<VOICES;++i) {
sig=0;
for (int j=0; j < OSCSPERVOICE; ++j) {
sig+=osc[i*OSCSPERVOICE + j].Process(); // sum oscillators in each voice
}
sig=sig/OSCSPERVOICE; // scale down by number of oscillators
if (StoredNotes[i] == -1) gate=false; // if voice is allocated then it is active
else gate=true;
envelope=env[i].Process(gate);
sig=sig*envelope;
filt[i].SetFreq(filterfreq+envelope*filtersweep);
//filt[i].SetFreq(200+envelope*3000*(lfo.Process()+1));
filtsig=filt[i].Process(sig);
outsig+=filtsig;
}
outsig=outsig*8/VOICES; // scale the sum
verb.Process(outsig, outsig, &wetvl, &wetvr);
out=outsig + wetvl*reverblevel;
// convert generated float value -1.0 to +1.0 to int16 used by Teensy Audio
int32_t val = out*MULT_16;
block->data[i] = val >> 16;
}
transmit(block);
release(block);
}
// teensy audio objects and patch creation
AudioOutputI2S out;
//AudioOutputUSB outUSB; // USB audio breaks up badly
AudioControlSGTL5000 audioShield;
AudioSynthDaisySP synth; // create the daisysp synth audio object
AudioConnection patchCord1(synth,0,out,0);
AudioConnection patchCord2(synth,0,out,1);
//AudioConnection patchCord3(synth,0,outUSB,0);
//AudioConnection patchCord4(synth,0,outUSB,1);
// frequencies for all 127 MIDI Note numbers.
// C C# D D# E F F# G G# A A# B
const float NoteNumToFreq[] = {
8.18, 8.66, 9.18, 9.72, 10.30, 10.91, 11.56, 12.25, 12.98, 13.75, 14.57, 15.43,
16.35, 17.32, 18.35, 19.45, 20.60, 21.83, 23.12, 24.50, 25.96, 27.50, 29.14, 30.87,
32.70, 34.65, 36.71, 38.89, 41.20, 43.65, 46.25, 49.00, 51.91, 55.00, 58.27, 61.74,
65.41, 69.30, 73.42, 77.78, 82.41, 87.31, 92.50, 98.00, 103.82, 110.00, 116.54, 123.47,
130.81, 138.59, 146.83, 155.56, 164.81, 174.61, 184.99, 195.99, 207.65, 220.00, 233.08, 246.94,
261.63, 277.18, 293.66, 311.13, 329.63, 349.23, 369.99, 391.99, 415.31, 440.00, 466.16, 493.88,
523.25, 554.37, 587.33, 622.25, 659.26, 698.46, 739.99, 783.99, 830.61, 880.00, 932.32, 987.77,
1046.50, 1108.73, 1174.66, 1244.51, 1318.51, 1396.91, 1479.98, 1567.98, 1661.22, 1760.00, 1864.66, 1975.53,
2093.00, 2217.46, 2349.32, 2489.02, 2637.02, 2793.83, 2959.96, 3135.96, 3322.44, 3520.00, 3729.31, 3951.07,
4186.01, 4434.92, 4698.64, 4978.03, 5274.04, 5587.65, 5919.91, 6271.93, 6644.88, 7040.00, 7458.62, 7902.13,
8372.02, 8869.84, 9397.27, 9956.06, 10548.08, 11175.30, 11839.82, 12543.85 };
void setup() {
Serial.begin(38400);
#ifdef DEBUG
while (!Serial) {
// wait for Arduino Serial Monitor to be ready
}
Serial.println("starting setup");
#endif
for (int i=0; i< VOICES;++i) {
StoredNotes[i]=-1; // initialize the note allocation array
for (int j=0; j< OSCSPERVOICE; ++j) {
osc[i*OSCSPERVOICE +j].Init(samplerate); // initialize the voice objects
osc[i*OSCSPERVOICE +j].SetWaveform(Oscillator::WAVE_POLYBLEP_SAW); // changing waveforms on the fly seems to cause a crash
}
env[i].Init(samplerate);
env[i].SetTime(ADENV_SEG_DECAY, 1.0f);
filt[i].Init(samplerate);
}
lfo.Init(samplerate); // Init LFO oscillator
lfo.SetFreq(0.1);
// env.SetCurve(-15.0f); // only for AR env
// initialize the reverb object and set its initial parameters
verb.Init(samplerate);
verb.SetFeedback(0.87);
verb.SetLpFreq(10000.0f);
// Enable the AudioShield
AudioMemory(10); // only uses 2 blocks
Serial.println("enabling audio shield");
audioShield.enable();
audioShield.volume(0.8);
// Handles for the USB MIDI callbacks
usbMIDI.setHandleNoteOn(myNoteOn);
usbMIDI.setHandleNoteOff(myNoteOff);
usbMIDI.setHandleControlChange(myControlChange);
usbMIDI.setHandleAfterTouchPoly(myAfterTouch);
#ifdef DEBUG
Serial.println("finished setup");
#endif
}
// Only looking for incoming MIDI events in the loop()
// myNoteOn(), myNoteOff() and myControlChange() will be processed on incoming MIDI messages.
void loop() {
usbMIDI.read();
#ifdef DEBUG
// DEBUG - Microcontroller Load Check
if (five_sec.check() == 1)
{
Serial.print("Proc = ");
Serial.print(AudioProcessorUsage());
Serial.print(" (");
Serial.print(AudioProcessorUsageMax());
Serial.print("), Mem = ");
Serial.print(AudioMemoryUsage());
Serial.print(" (");
Serial.print(AudioMemoryUsageMax());
Serial.println(")");
}
#endif
}
// Callback for incoming NoteOn messages
// Handling the voice allocation here.
void myNoteOn(byte channel, byte note, byte velocity) {
int i=0;
while( i < VOICES){
if (StoredNotes[i] == -1) { // if voice is idle
StoredNotes[i] = int(note); // allocate this voice
osc[i*OSCSPERVOICE].SetFreq(NoteNumToFreq[note]);
osc[i*OSCSPERVOICE+1].SetFreq(NoteNumToFreq[note]+detune); // quick and dirty detune
osc[i*OSCSPERVOICE+2].SetFreq(NoteNumToFreq[note]-detune);
// env[i].Trigger(); // ADSR triggering happens in the sample loop
break;
}
++i;
}
}
// Callback for incoming NoteOff messages
// Releasing voices to be re-allocated here.
void myNoteOff(byte channel, byte note, byte velocity) {
for (int i=0; i < VOICES; ++i){
int k = int(note);
if (StoredNotes[i] == k) { // if this voice matches the note we are silencing
StoredNotes[i] = -1; // deallocate the voice
}
}
}
// Callback for incoming CC messages
// I'm using an external MIDI controller (Arturia Beatstep) to set voice parameters
// you can also do this with pots and AnalogRead()
void myControlChange(byte channel, byte control, byte value) {
float val = float(value) / 127; // convert to 0-1
for (int i=0; i < VOICES; ++i){
switch (control) {
case 101:
waveform=value/40; // chaning waveforms on the fly doesn't seem to work
break;
case 102:
detune=val*5; // oscillator detune
break;
case 103:
break;
case 105:
env[i].SetTime(ADSR_SEG_ATTACK,val);
break;
case 106:
env[i].SetTime(ADSR_SEG_DECAY,val);
break;
case 107:
env[i].SetSustainLevel(val);
break;
case 108:
env[i].SetTime(ADSR_SEG_RELEASE,val);
break;
case 113:
filterfreq=50+val*2000; // filter cutoff
break;
case 114:
filtersweep=val*10000; // filter sweep - controlled by envelope
break;
case 115:
filt[i].SetRes(val); // filter resonance
break;
case 116:
reverblevel=val; // reverb
break;
default:
break;
}
}
}
// Callback for incoming Aftertouch messages
void myAfterTouch(byte channel, byte note, byte value) {
float val = float(value) / 127; // convert to 0-1
}
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