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/*
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Oscillator with Pitch modulation |
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Created: Patrick Radius, Jan 2017 |
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Purpose: Demonstrates the ability of the oscillator and pitch modulation. |
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Demonstrates audio processing using floating point data type. |
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Uses Teensy Audio Adapter. |
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MIT License. use at your own risk. |
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*/ |
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//These are the includes from the Teensy Audio Library
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#include <Audio.h> //Teensy Audio Librarya |
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#include <Wire.h> |
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#include <SPI.h> |
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#include <SD.h> |
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#include <SerialFlash.h> |
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#include <OpenAudio_ArduinoLibrary.h> //for AudioConvert_I16toF32, AudioConvert_F32toI16, and AudioEffectGain_F32 |
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#define DO_USB 1 //set to 1 to enable USB audio. Be sure to go under the "Tools" menu and do "USB Type" -> "Audio"
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//create audio library objects for handling the audio
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AudioControlSGTL5000_Extended sgtl5000; //controller for the Teensy Audio Board
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AudioOutputI2S i2s_out; //Digital audio *to* the Teensy Audio Board DAC. Expects Int16. Stereo
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AudioConvert_F32toI16 float2Int; //Converts Float to Int16. See class in AudioStream_F32.h
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AudioSynthWaveform_F32 osc1; // Audio-rate oscillator.
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AudioSynthWaveform_F32 lfo1; // Low-frequency oscillator to modulate the main oscillator with.
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AudioConnection_F32 patchCord1(osc1, 0, float2Int, 0); // connect the oscillator to the float 2 int converter
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AudioConnection_F32 patchCord2(lfo1, 0, osc1, 0); // connect the output of the lfo to the mod input of the oscillator
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AudioConnection patchCord3(float2Int, 0, i2s_out, 0); //Left out.
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AudioConnection patchCord4(float2Int, 0, i2s_out, 1); //Right out.
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//For output, you always need an i2s or else you have no clock to drive the audio subsystem.
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//So, even if you're not going to use the headphone/line-out, you need these lines
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AudioConnection patchCord5(float2Int, 0, i2s_out, 0); //connect the Left float processor to the Left output.
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AudioConnection patchCord6(float2Int, 0, i2s_out, 1); //connect the Right float processor to the Right output.
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//Now, if you want USB output, it gets enabled here
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#if DO_USB |
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AudioOutputUSB usb_out; |
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AudioConnection patchCord7(float2Int, 0, usb_out, 0); //connect the float processor to the Left output
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AudioConnection patchCord8(float2Int, 0, usb_out, 1); //connect the float processor to the Right output
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#endif |
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// define the overall setup() function, the function that is called once when the device is booting
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void setup() { |
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Serial.begin(115200); //open the USB serial link to enable debugging messages
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delay(500); //give the computer's USB serial system a moment to catch up.
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Serial.println("Teensy: AudioSynthWaveform_F32 example..."); //identify myself over the USB serial
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// Audio connections require memory, and the record queue
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// uses this memory to buffer incoming audio.
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AudioMemory(12); //allocate Int16 audio data blocks
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AudioMemory_F32(10); //allocate Float32 audio data blocks
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// Enable the audio shield, no input, and enable output
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sgtl5000.enable(); //start the audio board
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sgtl5000.volume(0.5); //volume can be 0.0 to 1.0. 0.5 seems to be the usual default.
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osc1.begin(1.0, 440.0, 0); // run main oscillator at nominal amplitude, note A4, sine wave
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lfo1.begin(1.0, 0.25, 1); // run lfo at nominal amplitude, 0.25hz (so 4 second period), saw wave
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osc1.pitchModAmount(3.0); // Set the amount of pitch modulation to be around 3 octaves
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} //end setup()
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unsigned long curTime_millis = 0; |
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unsigned long lastMemUpdate_millis=0; |
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// define the loop() function, the function that is repeated over and over for the life of the device
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void loop() { |
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printCPUandMemoryUsage(&Serial); |
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} //end loop();
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void printCPUandMemoryUsage(Stream *s) { |
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//print status information to the Serial port
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curTime_millis = millis(); //what time is it right now
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if ((curTime_millis - lastMemUpdate_millis) < 0) lastMemUpdate_millis=0; //handle case where millis wraps around!
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if ((curTime_millis - lastMemUpdate_millis) > 2000) { // print a summary of the current & maximum usage
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s->print("Usage/Max: "); |
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s->print("oscillator CPU = "); s->print(osc1.processorUsage()); s->print("/"); s->print(osc1.processorUsageMax());s->print(", "); |
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s->print("LFO CPU = "); s->print(lfo1.processorUsage()); s->print("/"); s->print(lfo1.processorUsageMax());s->print(", "); |
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s->print("all CPU = " ); s->print(AudioProcessorUsage()); s->print("/"); s->print(AudioProcessorUsageMax());s->print(", "); |
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s->print("Int16 Mem = ");s->print(AudioMemoryUsage()); s->print("/"); s->print(AudioMemoryUsageMax());s->print(", "); |
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s->print("Float Mem = ");s->print(AudioMemoryUsage_F32());s->print("/"); s->print(AudioMemoryUsageMax_F32()); s->print(", "); |
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s->println(); |
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lastMemUpdate_millis = curTime_millis; //we will use this value the next time around.
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}
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}; |
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@ -0,0 +1,92 @@ |
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#include <arm_math.h> |
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#include "synth_waveform_F32.h" |
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void AudioSynthWaveform_F32::update(void) { |
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audio_block_f32_t *block, *lfo; |
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if (_magnitude == 0.0f) return; |
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block = allocate_f32(); |
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if (!block) return; |
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lfo = receiveReadOnly_f32(0); |
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switch (_OscillatorMode) { |
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case OSCILLATOR_MODE_SINE: |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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applyMod(i, lfo); |
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block->data[i] = arm_sin_f32(_Phase); |
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_Phase += _PhaseIncrement; |
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while (_Phase >= twoPI) { |
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_Phase -= twoPI; |
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} |
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} |
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break; |
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case OSCILLATOR_MODE_SAW: |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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applyMod(i, lfo); |
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block->data[i] = 1.0f - (2.0f * _Phase / twoPI); |
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_Phase += _PhaseIncrement; |
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while (_Phase >= twoPI) { |
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_Phase -= twoPI; |
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} |
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} |
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break; |
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case OSCILLATOR_MODE_SQUARE: |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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applyMod(i, lfo); |
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if (_Phase <= _PI) { |
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block->data[i] = 1.0f; |
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} else { |
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block->data[i] = -1.0f; |
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} |
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_Phase += _PhaseIncrement; |
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while (_Phase >= twoPI) { |
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_Phase -= twoPI; |
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} |
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} |
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break; |
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case OSCILLATOR_MODE_TRIANGLE: |
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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applyMod(i, lfo); |
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float32_t value = -1.0f + (2.0f * _Phase / twoPI); |
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block->data[i] = 2.0f * (fabs(value) - 0.5f); |
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_Phase += _PhaseIncrement; |
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while (_Phase >= twoPI) { |
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_Phase -= twoPI; |
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} |
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} |
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break; |
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} |
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if (_magnitude != 1.0f) { |
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arm_scale_f32(block->data, _magnitude, block->data, AUDIO_BLOCK_SAMPLES); |
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} |
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if (lfo) { |
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release(lfo); |
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} |
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AudioStream_F32::transmit(block); |
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AudioStream_F32::release(block); |
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} |
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inline float32_t AudioSynthWaveform_F32::applyMod(uint32_t sample, audio_block_f32_t *lfo) { |
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if (_PortamentoSamples > 0 && _CurrentPortamentoSample++ < _PortamentoSamples) { |
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_Frequency+=_PortamentoIncrement; |
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} |
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float32_t osc_frequency = _Frequency; |
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if (lfo && _PitchModAmt > 0.0f) { |
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osc_frequency = _Frequency * powf(2.0f, 0.0f / 1200.0f + lfo->data[sample] * _PitchModAmt); |
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} |
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_PhaseIncrement = osc_frequency * twoPI / AUDIO_SAMPLE_RATE_EXACT; |
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return osc_frequency; |
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} |
@ -0,0 +1,118 @@ |
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/*
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* AudioSynthWaveform_F32 |
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*
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* Created: Patrick Radius, January 2017 |
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* Purpose: Generate waveforms at a given frequency and amplitude. Allows for pitch-modulation and portamento. |
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*
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* This processes a single stream fo audio data (ie, it is mono)
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*
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* MIT License. use at your own risk. |
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*/ |
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#ifndef SYNTHWAVEFORMF32_H |
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#define SYNTHWAVEFORMF32_H |
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#include <arm_math.h> |
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#include <AudioStream_F32.h> |
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class AudioSynthWaveform_F32 : public AudioStream_F32 |
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{ |
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public: |
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enum OscillatorMode { |
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OSCILLATOR_MODE_SINE = 0, |
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OSCILLATOR_MODE_SAW, |
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OSCILLATOR_MODE_SQUARE, |
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OSCILLATOR_MODE_TRIANGLE |
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}; |
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AudioSynthWaveform_F32(void) : AudioStream_F32(1, inputQueueArray_f32), |
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_PI(2*acos(0.0f)), |
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twoPI(2 * _PI), |
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_OscillatorMode(OSCILLATOR_MODE_SINE), |
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_Frequency(440.0f), |
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_Phase(0.0f), |
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_PhaseIncrement(0.0f), |
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_PitchModAmt(0.0f), |
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_PortamentoIncrement(0.0f), |
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_PortamentoSamples(0), |
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_CurrentPortamentoSample(0), |
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_NotesPlaying(0) {}; |
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void frequency(float32_t freq) { |
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float32_t nyquist = AUDIO_SAMPLE_RATE_EXACT/2; |
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if (freq < 0.0) freq = 0.0; |
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else if (freq > nyquist) freq = nyquist; |
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if (_PortamentoSamples > 0 && _NotesPlaying > 0) { |
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_PortamentoIncrement = (freq - _Frequency) / (float32_t)_PortamentoSamples; |
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_CurrentPortamentoSample = 0; |
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} else { |
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_Frequency = freq; |
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} |
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_PhaseIncrement = _Frequency * twoPI / AUDIO_SAMPLE_RATE_EXACT; |
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} |
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void amplitude(float32_t n) { |
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if (n < 0) n = 0; |
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_magnitude = n; |
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} |
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void begin(short t_type) { |
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_Phase = 0; |
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oscillatorMode(t_type); |
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} |
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void begin(float32_t t_amp, float32_t t_freq, short t_type) { |
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amplitude(t_amp); |
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frequency(t_freq); |
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begin(t_type); |
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} |
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void pitchModAmount(float32_t amount) { |
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_PitchModAmt = amount; |
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} |
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void oscillatorMode(int mode) { |
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_OscillatorMode = (OscillatorMode)mode; |
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} |
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void portamentoTime(float32_t slidetime) { |
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_PortamentoTime = slidetime; |
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_PortamentoSamples = floorf(slidetime * AUDIO_SAMPLE_RATE_ROUNDED); |
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} |
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void onNoteOn() { |
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_NotesPlaying++; |
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} |
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void onNoteOff() { |
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if (_NotesPlaying > 0) { |
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_NotesPlaying--; |
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} |
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} |
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void update(void); |
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private: |
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inline float32_t applyMod(uint32_t sample, audio_block_f32_t *lfo); |
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const float32_t _PI; |
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float32_t twoPI; |
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OscillatorMode _OscillatorMode; |
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float32_t _Frequency; |
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float32_t _Phase; |
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float32_t _PhaseIncrement; |
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float32_t _magnitude; |
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float32_t _PitchModAmt; |
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float32_t _PortamentoTime; |
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float32_t _PortamentoIncrement; |
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uint64_t _PortamentoSamples; |
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uint64_t _CurrentPortamentoSample; |
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uint8_t _NotesPlaying; |
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audio_block_f32_t *inputQueueArray_f32[1]; |
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}; |
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#endif |
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