Fixed voice allocation code in Modal example. Added polysynth example.main
Rich Heslip
4 years ago
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6aa5a65294
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// test of DaisySP synth object for the Teensy audio library
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// simple poly synth
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// some of this code was cribbed from the Faust for Teensy Additivesynth example
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// RH March 29 2021
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#include <Audio.h> |
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#include <Metro.h> |
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//#define DEBUG // comment out to remove debug code
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#ifdef DEBUG |
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Metro five_sec=Metro(5000); // Set up a 5 second Metro for performance stats
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#endif |
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// constants for integer to float and float to integer conversion
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#define MULT_16 2147483647 |
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#define DIV_16 4.6566129e-10 |
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// for polyphony - an array of all current notes.
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// Value -1 means the note is off (not sounding).
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#define VOICES 16 //
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int StoredNotes[VOICES]; |
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#include "daisysp.h" |
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using namespace daisysp; |
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// including the source files is a pain but that way you compile in only the modules you need
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// DaisySP statically allocates memory and some modules e.g. reverb use a lot of ram
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#include "synthesis/oscillator.cpp" |
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#include "control/adsr.cpp" |
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#include "filters/moogladder.cpp" |
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#include "effects/reverbsc.cpp" // uses a LOT of ram |
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float samplerate=AUDIO_SAMPLE_RATE_EXACT; |
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// parameters we can modify via MIDI CCs
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int waveform=0; |
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float detune=0; |
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float filterfreq=100; |
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float filtersweep=3000; |
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float filterresonance=0.3; |
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float reverblevel=0.1; |
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float lfofreq=0.1; |
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float lfofreqdepth=0; |
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float lfofilterdepth=0; |
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// create daisySP processing objects
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#define OSCSPERVOICE 3 // note - the detune code is set up for 3 oscillators
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Oscillator osc[VOICES * OSCSPERVOICE]; |
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Oscillator lfo; |
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Adsr env[VOICES]; |
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MoogLadder filt[VOICES]; |
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ReverbSc verb;
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// this is the function called by the AudioSynthDaisySP object when it needs a block of samples
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void AudioSynthDaisySP::update(void) |
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{ |
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float out,sig,outsig,envelope,filtsig,wetvl, wetvr; |
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bool gate; |
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audio_block_t *block; |
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block = allocate(); // grab an audio block
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if (!block) { |
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return; |
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} |
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for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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//**** insert daisySP generators here
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outsig=0; // sum up voices
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for (int i=0; i<VOICES;++i) { |
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sig=0; |
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for (int j=0; j < OSCSPERVOICE; ++j) { |
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sig+=osc[i*OSCSPERVOICE + j].Process(); // sum oscillators in each voice
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} |
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sig=sig/OSCSPERVOICE; // scale down by number of oscillators
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if (StoredNotes[i] == -1) gate=false; // if voice is allocated then it is active
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else gate=true;
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envelope=env[i].Process(gate); |
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sig=sig*envelope; |
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filt[i].SetFreq(filterfreq+envelope*filtersweep); |
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//filt[i].SetFreq(200+envelope*3000*(lfo.Process()+1));
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filtsig=filt[i].Process(sig); |
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outsig+=filtsig; |
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} |
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outsig=outsig*8/VOICES; // scale the sum
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verb.Process(outsig, outsig, &wetvl, &wetvr);
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out=outsig + wetvl*reverblevel; |
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// convert generated float value -1.0 to +1.0 to int16 used by Teensy Audio
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int32_t val = out*MULT_16; |
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block->data[i] = val >> 16; |
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} |
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transmit(block); |
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release(block); |
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} |
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// teensy audio objects and patch creation
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AudioOutputI2S out; |
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//AudioOutputUSB outUSB; // USB audio breaks up badly
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AudioControlSGTL5000 audioShield; |
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AudioSynthDaisySP synth; // create the daisysp synth audio object
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AudioConnection patchCord1(synth,0,out,0); |
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AudioConnection patchCord2(synth,0,out,1); |
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//AudioConnection patchCord3(synth,0,outUSB,0);
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//AudioConnection patchCord4(synth,0,outUSB,1);
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// frequencies for all 127 MIDI Note numbers.
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// C C# D D# E F F# G G# A A# B
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const float NoteNumToFreq[] = { |
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8.18, 8.66, 9.18, 9.72, 10.30, 10.91, 11.56, 12.25, 12.98, 13.75, 14.57, 15.43,
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16.35, 17.32, 18.35, 19.45, 20.60, 21.83, 23.12, 24.50, 25.96, 27.50, 29.14, 30.87,
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32.70, 34.65, 36.71, 38.89, 41.20, 43.65, 46.25, 49.00, 51.91, 55.00, 58.27, 61.74,
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65.41, 69.30, 73.42, 77.78, 82.41, 87.31, 92.50, 98.00, 103.82, 110.00, 116.54, 123.47,
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130.81, 138.59, 146.83, 155.56, 164.81, 174.61, 184.99, 195.99, 207.65, 220.00, 233.08, 246.94,
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261.63, 277.18, 293.66, 311.13, 329.63, 349.23, 369.99, 391.99, 415.31, 440.00, 466.16, 493.88,
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523.25, 554.37, 587.33, 622.25, 659.26, 698.46, 739.99, 783.99, 830.61, 880.00, 932.32, 987.77,
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1046.50, 1108.73, 1174.66, 1244.51, 1318.51, 1396.91, 1479.98, 1567.98, 1661.22, 1760.00, 1864.66, 1975.53,
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2093.00, 2217.46, 2349.32, 2489.02, 2637.02, 2793.83, 2959.96, 3135.96, 3322.44, 3520.00, 3729.31, 3951.07,
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4186.01, 4434.92, 4698.64, 4978.03, 5274.04, 5587.65, 5919.91, 6271.93, 6644.88, 7040.00, 7458.62, 7902.13,
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8372.02, 8869.84, 9397.27, 9956.06, 10548.08, 11175.30, 11839.82, 12543.85 }; |
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void setup() {
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Serial.begin(38400); |
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#ifdef DEBUG |
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while (!Serial) { |
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// wait for Arduino Serial Monitor to be ready
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} |
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Serial.println("starting setup");
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#endif |
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for (int i=0; i< VOICES;++i) {
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StoredNotes[i]=-1; // initialize the note allocation array
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for (int j=0; j< OSCSPERVOICE; ++j) { |
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osc[i*OSCSPERVOICE +j].Init(samplerate); // initialize the voice objects
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osc[i*OSCSPERVOICE +j].SetWaveform(Oscillator::WAVE_POLYBLEP_SAW); // changing waveforms on the fly seems to cause a crash
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} |
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env[i].Init(samplerate); |
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env[i].SetTime(ADENV_SEG_DECAY, 1.0f); |
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filt[i].Init(samplerate); |
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} |
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lfo.Init(samplerate); // Init LFO oscillator
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lfo.SetFreq(0.1); |
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// env.SetCurve(-15.0f); // only for AR env
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// initialize the reverb object and set its initial parameters
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verb.Init(samplerate); |
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verb.SetFeedback(0.87); |
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verb.SetLpFreq(10000.0f); |
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// Enable the AudioShield
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AudioMemory(10); // only uses 2 blocks
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Serial.println("enabling audio shield");
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audioShield.enable(); |
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audioShield.volume(0.8); |
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// Handles for the USB MIDI callbacks
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usbMIDI.setHandleNoteOn(myNoteOn); |
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usbMIDI.setHandleNoteOff(myNoteOff); |
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usbMIDI.setHandleControlChange(myControlChange); |
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usbMIDI.setHandleAfterTouchPoly(myAfterTouch); |
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#ifdef DEBUG |
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Serial.println("finished setup");
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#endif |
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} |
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// Only looking for incoming MIDI events in the loop()
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// myNoteOn(), myNoteOff() and myControlChange() will be processed on incoming MIDI messages.
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void loop() { |
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usbMIDI.read(); |
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#ifdef DEBUG |
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// DEBUG - Microcontroller Load Check
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if (five_sec.check() == 1) |
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{ |
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Serial.print("Proc = "); |
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Serial.print(AudioProcessorUsage()); |
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Serial.print(" (");
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Serial.print(AudioProcessorUsageMax()); |
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Serial.print("), Mem = "); |
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Serial.print(AudioMemoryUsage()); |
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Serial.print(" (");
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Serial.print(AudioMemoryUsageMax()); |
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Serial.println(")"); |
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} |
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#endif |
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} |
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// Callback for incoming NoteOn messages
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// Handling the voice allocation here.
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void myNoteOn(byte channel, byte note, byte velocity) { |
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int i=0; |
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while( i < VOICES){ |
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if (StoredNotes[i] == -1) { // if voice is idle
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StoredNotes[i] = int(note); // allocate this voice
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osc[i*OSCSPERVOICE].SetFreq(NoteNumToFreq[note]);
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osc[i*OSCSPERVOICE+1].SetFreq(NoteNumToFreq[note]+detune); // quick and dirty detune
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osc[i*OSCSPERVOICE+2].SetFreq(NoteNumToFreq[note]-detune); |
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// env[i].Trigger(); // ADSR triggering happens in the sample loop
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break; |
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} |
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++i; |
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}
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} |
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// Callback for incoming NoteOff messages
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// Releasing voices to be re-allocated here.
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void myNoteOff(byte channel, byte note, byte velocity) { |
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for (int i=0; i < VOICES; ++i){ |
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int k = int(note);
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if (StoredNotes[i] == k) { // if this voice matches the note we are silencing
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StoredNotes[i] = -1; // deallocate the voice
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} |
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} |
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} |
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// Callback for incoming CC messages
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// I'm using an external MIDI controller (Arturia Beatstep) to set voice parameters
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// you can also do this with pots and AnalogRead()
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void myControlChange(byte channel, byte control, byte value) { |
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float val = float(value) / 127; // convert to 0-1
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for (int i=0; i < VOICES; ++i){ |
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switch (control) { |
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case 101: |
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waveform=value/40; // chaning waveforms on the fly doesn't seem to work
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break; |
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case 102: |
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detune=val*5; // oscillator detune
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break; |
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case 103: |
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break; |
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case 105: |
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env[i].SetTime(ADSR_SEG_ATTACK,val); |
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break; |
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case 106: |
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env[i].SetTime(ADSR_SEG_DECAY,val); |
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break;
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case 107: |
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env[i].SetSustainLevel(val); |
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break;
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case 108: |
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env[i].SetTime(ADSR_SEG_RELEASE,val); |
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break; |
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case 113: |
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filterfreq=50+val*2000; // filter cutoff
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break; |
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case 114: |
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filtersweep=val*10000; // filter sweep - controlled by envelope
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break; |
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case 115: |
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filt[i].SetRes(val); // filter resonance
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break; |
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case 116: |
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reverblevel=val; // reverb
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break; |
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default: |
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break; |
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} |
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} |
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} |
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// Callback for incoming Aftertouch messages
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void myAfterTouch(byte channel, byte note, byte value) { |
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float val = float(value) / 127; // convert to 0-1
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} |
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