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329 lines
12 KiB
329 lines
12 KiB
/*
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* AudioEffectAnalogChorus.cpp
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*
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* Created on: Jan 7, 2018
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* Author: slascos
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*/
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#include <new>
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#include <cmath>
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#include "AudioEffectAnalogChorusFilters.h"
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#include "AudioEffectAnalogChorus.h"
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using namespace BALibrary;
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namespace BAEffects {
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constexpr int MIDI_CHANNEL = 0;
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constexpr int MIDI_CONTROL = 1;
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constexpr float DELAY_REFERENCE_F = static_cast<float>(AUDIO_BLOCK_SAMPLES/2);
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AudioEffectAnalogChorus::AudioEffectAnalogChorus()
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: AudioStream(1, m_inputQueueArray)
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{
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m_memory = new AudioDelay(m_DEFAULT_AVERAGE_DELAY_MS + m_DELAY_RANGE);
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m_maxDelaySamples = calcAudioSamples(m_DEFAULT_AVERAGE_DELAY_MS + m_DELAY_RANGE);
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m_averageDelaySamples = static_cast<float>(calcAudioSamples(m_DEFAULT_AVERAGE_DELAY_MS));
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m_delayRange = static_cast<float>(calcAudioSamples(m_DELAY_RANGE));
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m_constructFilter();
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m_lfo.setWaveform(Waveform::TRIANGLE);
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m_lfo.setRateAudio(4.0f); // Default to 4 Hz
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}
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AudioEffectAnalogChorus::~AudioEffectAnalogChorus()
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{
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if (m_memory) delete m_memory;
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if (m_iir) delete m_iir;
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}
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// This function just sets up the default filter and coefficients
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void AudioEffectAnalogChorus::m_constructFilter(void)
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{
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// Use CE2 coefficients by default
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m_iir = new IirBiQuadFilterHQ(CE2_NUM_STAGES, reinterpret_cast<const int32_t *>(&CE2), CE2_COEFF_SHIFT);
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}
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void AudioEffectAnalogChorus::setWaveform(BALibrary::Waveform waveform)
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{
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switch(waveform) {
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case Waveform::SINE :
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case Waveform::TRIANGLE :
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case Waveform::SAWTOOTH :
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m_lfo.setWaveform(waveform);
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break;
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default :
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Serial.println("AudioEffectAnalogChorus::setWaveform: Unsupported Waveform");
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}
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}
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void AudioEffectAnalogChorus::setFilterCoeffs(int numStages, const int32_t *coeffs, int coeffShift)
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{
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m_iir->changeFilterCoeffs(numStages, coeffs, coeffShift);
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}
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void AudioEffectAnalogChorus::setFilter(Filter filter)
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{
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switch(filter) {
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case Filter::WARM :
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m_iir->changeFilterCoeffs(WARM_NUM_STAGES, reinterpret_cast<const int32_t *>(&WARM), WARM_COEFF_SHIFT);
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break;
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case Filter::DARK :
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m_iir->changeFilterCoeffs(DARK_NUM_STAGES, reinterpret_cast<const int32_t *>(&DARK), DARK_COEFF_SHIFT);
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break;
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case Filter::CE2 :
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default:
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m_iir->changeFilterCoeffs(CE2_NUM_STAGES, reinterpret_cast<const int32_t *>(&CE2), CE2_COEFF_SHIFT);
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break;
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}
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}
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void AudioEffectAnalogChorus::update(void)
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{
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audio_block_t *inputAudioBlock = receiveReadOnly(); // get the next block of input samples
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// Check is block is disabled
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if (m_enable == false) {
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// do not transmit or proess any audio, return as quickly as possible.
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if (inputAudioBlock) release(inputAudioBlock);
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// release all held memory resources
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if (m_previousBlock) {
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release(m_previousBlock); m_previousBlock = nullptr;
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}
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if (!m_externalMemory) {
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// when using internal memory we have to release all references in the ring buffer
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while (m_memory->getRingBuffer()->size() > 0) {
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audio_block_t *releaseBlock = m_memory->getRingBuffer()->front();
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m_memory->getRingBuffer()->pop_front();
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if (releaseBlock) release(releaseBlock);
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}
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}
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return;
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}
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// Check is block is bypassed, if so either transmit input directly or create silence
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if (m_bypass == true) {
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// transmit the input directly
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if (!inputAudioBlock) {
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// create silence
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inputAudioBlock = allocate();
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if (!inputAudioBlock) { return; } // failed to allocate
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else {
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clearAudioBlock(inputAudioBlock);
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}
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}
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transmit(inputAudioBlock, 0);
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release(inputAudioBlock);
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return;
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}
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// Otherwise perform normal processing
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// In order to make use of the SPI DMA, we need to request the read from memory first,
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// then do other processing while it fills in the back.
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audio_block_t *blockToOutput = nullptr; // this will hold the output audio
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blockToOutput = allocate();
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if (!blockToOutput) return; // skip this update cycle due to failure
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// get the data. If using external memory with DMA, this won't be filled until
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// later.
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// We need to grab two blocks of audio since the modulating delay value from the LFO
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// can exceed the length of one audio block during the time frame of one audio block.
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int16_t extendedBuffer[(2*AUDIO_BLOCK_SAMPLES)]; // need one more sample for interpolating between 128th and 129th (last sample)
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// Get next vector of lfo values, they will range range from -1.0 to +1.0f.
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float *lfoValues = m_lfo.getNextVector();
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//float lfoValues[128];
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for (int i=0; i<128; i++) { lfoValues[i] = lfoValues[i] * m_lfoDepth; }
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// Calculate the starting delay from the first lfo sample. This will represent the 'reference' delay
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// for this output block
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float referenceDelay = m_averageDelaySamples + (lfoValues[0] * m_delayRange);
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unsigned delaySamples = static_cast<unsigned>(referenceDelay); // round down to the nearest audio sample for indexing into AudioDelay class
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// From a given current delay value, while reading out the next 128, the delay could slew up or down
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// AUDIO_BLOCK_SAMPLES/2 cycles of delay. For example...
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// Pitching up : current + 128 + 64
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// Pitching down: current - 64 + 128
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// We need to grab 2*AUDIO_BLOCK_SAMPLES. Be aware that audio samples are stored BACKWARDS in the buffers.
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// m_memory->getSamples(extendedBuffer , delaySamples - (AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
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// m_memory->getSamples(extendedBuffer + AUDIO_BLOCK_SAMPLES, delaySamples +( AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
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m_memory->getSamples(extendedBuffer + AUDIO_BLOCK_SAMPLES, delaySamples - (AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
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m_memory->getSamples(extendedBuffer , delaySamples +( AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
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// If using DMA, we need something else to do while that read executes, so
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// move on to input preprocessing
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// Preprocessing
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audio_block_t *preProcessed = allocate();
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// mix the input with the feedback path in the pre-processing stage
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m_preProcessing(preProcessed, inputAudioBlock, m_previousBlock);
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// consider doing the BBD post processing here to use up more time while waiting
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// for the read data to come back
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audio_block_t *blockToRelease = m_memory->addBlock(preProcessed);
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// BACK TO OUTPUT PROCESSING
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double bufferIndexFloat;
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int delayIndex;
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for (int i=0, j=AUDIO_BLOCK_SAMPLES-1; i<AUDIO_BLOCK_SAMPLES; i++,j--) {
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// each output sample will be an interpolated value between two samples
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// the precise delay value will be based on the LFO vector values.
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// For each output sample, calculate the floating point delay offset from the reference delay.
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// This will be an offset from location AUDIO_BLOCK_SAMPLES/2 (e.g. 64) in the buffer.
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float offsetDelayFromRef = m_averageDelaySamples + (lfoValues[i] * m_delayRange) - referenceDelay;
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float bufferPosition = DELAY_REFERENCE_F + offsetDelayFromRef;
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// Get the interpolation coefficients from the fractional part of the buffer position
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float fraction1 = modf(bufferPosition, &bufferIndexFloat);
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float fraction2 = 1.0f - fraction1;
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//fraction1 = 0.5f;
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//fraction2 = 0.5f;
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delayIndex = static_cast<unsigned>(bufferIndexFloat);
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if ( (delayIndex < 0) || (delayIndex > 256) ) {
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Serial.println(String("lfoValues[") + i + String("]:") + lfoValues[i] +
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String(" referenceDelay:") + referenceDelay +
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String(" bufferPosition:") + bufferPosition +
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String(" delayIndex:") + delayIndex) ;
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}
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//delayIndex = 64+i;
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blockToOutput->data[j] = static_cast<int16_t>(
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(static_cast<float>(extendedBuffer[j+delayIndex]) * fraction1) +
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(static_cast<float>(extendedBuffer[j+delayIndex+1]) * fraction2) );
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//blockToOutput->data[i] = extendedBuffer[64+i];
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}
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// perform the wet/dry mix mix
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m_postProcessing(blockToOutput, inputAudioBlock, blockToOutput);
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transmit(blockToOutput);
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release(inputAudioBlock);
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release(m_previousBlock);
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m_previousBlock = blockToOutput;
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// if (m_externalMemory && m_memory->getSlot()->isUseDma()) {
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// // Using DMA
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// if (m_blockToRelease) release(m_blockToRelease);
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// m_blockToRelease = blockToRelease;
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// }
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if (m_blockToRelease) release(m_blockToRelease);
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m_blockToRelease = blockToRelease;
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}
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void AudioEffectAnalogChorus::m_preProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet)
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{
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memcpy(out->data, dry->data, sizeof(int16_t) * AUDIO_BLOCK_SAMPLES);
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// TODO: Clean this up with proper preprocessing
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// if ( out && dry && wet) {
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// alphaBlend(out, dry, wet, m_feedback);
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// m_iir->process(out->data, out->data, AUDIO_BLOCK_SAMPLES);
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// } else if (dry) {
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// memcpy(out->data, dry->data, sizeof(int16_t) * AUDIO_BLOCK_SAMPLES);
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// }
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}
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void AudioEffectAnalogChorus::m_postProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet)
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{
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if (!out) return; // no valid output buffer
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if ( out && dry && wet) {
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// Simulate the LPF IIR nature of the analog systems
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//m_iir->process(wet->data, wet->data, AUDIO_BLOCK_SAMPLES);
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alphaBlend(out, dry, wet, m_mix);
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} else if (dry) {
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memcpy(out->data, dry->data, sizeof(int16_t) * AUDIO_BLOCK_SAMPLES);
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}
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// Set the output volume
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gainAdjust(out, out, m_volume, 1);
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}
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void AudioEffectAnalogChorus::setDelayConfig(float averageDelayMs, float delayRangeMs)
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{
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setDelayConfig(calcAudioSamples(averageDelayMs), calcAudioSamples(delayRangeMs));
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}
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void AudioEffectAnalogChorus::setDelayConfig(size_t averageDelayNumSamples, size_t delayRangeNumSamples)
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{
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size_t delaySamples = averageDelayNumSamples + delayRangeNumSamples;
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m_averageDelaySamples = averageDelayNumSamples;
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m_delayRange = delayRangeNumSamples;
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if (delaySamples > m_memory->getMaxDelaySamples()) {
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// this exceeds max delay value, limit it.
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delaySamples = m_memory->getMaxDelaySamples();
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m_averageDelaySamples = delaySamples/2;
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m_delayRange = delaySamples/2;
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}
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if (!m_memory) { Serial.println("delay(): m_memory is not valid"); }
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}
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void AudioEffectAnalogChorus::rate(float rate)
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{
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// update the LFO by mapping the rate into the MIN/MAX range, pass to LFO in milliseconds
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m_lfo.setRateAudio(m_LFO_MIN_RATE + (rate * m_LFO_RANGE));
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}
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void AudioEffectAnalogChorus::processMidi(int channel, int control, int value)
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{
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float val = (float)value / 127.0f;
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if ((m_midiConfig[RATE][MIDI_CHANNEL] == channel) &&
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(m_midiConfig[RATE][MIDI_CONTROL] == control)) {
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// Rate
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Serial.println(String("AudioEffectAnalogChorus::rate: ") + 100*val + String("%"));
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rate(val);
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return;
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}
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if ((m_midiConfig[BYPASS][MIDI_CHANNEL] == channel) &&
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(m_midiConfig[BYPASS][MIDI_CONTROL] == control)) {
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// Bypass
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if (value >= 65) { bypass(false); Serial.println(String("AudioEffectAnalogChorus::not bypassed -> ON") + value); }
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else { bypass(true); Serial.println(String("AudioEffectAnalogChorus::bypassed -> OFF") + value); }
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return;
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}
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if ((m_midiConfig[DEPTH][MIDI_CHANNEL] == channel) &&
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(m_midiConfig[DEPTH][MIDI_CONTROL] == control)) {
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// depth
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Serial.println(String("AudioEffectAnalogChorus::depth: ") + 100*val + String("%"));
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depth(val);
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return;
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}
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if ((m_midiConfig[MIX][MIDI_CHANNEL] == channel) &&
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(m_midiConfig[MIX][MIDI_CONTROL] == control)) {
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// Mix
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Serial.println(String("AudioEffectAnalogChorus::mix: Dry: ") + 100*(1-val) + String("% Wet: ") + 100*val );
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mix(val);
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return;
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}
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if ((m_midiConfig[VOLUME][MIDI_CHANNEL] == channel) &&
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(m_midiConfig[VOLUME][MIDI_CONTROL] == control)) {
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// Volume
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Serial.println(String("AudioEffectAnalogChorus::volume: ") + 100*val + String("%"));
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volume(val);
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return;
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}
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}
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void AudioEffectAnalogChorus::mapMidiControl(int parameter, int midiCC, int midiChannel)
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{
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if (parameter >= NUM_CONTROLS) {
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return ; // Invalid midi parameter
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}
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m_midiConfig[parameter][MIDI_CHANNEL] = midiChannel;
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m_midiConfig[parameter][MIDI_CONTROL] = midiCC;
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}
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}
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