|
|
@ -18,29 +18,54 @@ constexpr unsigned NUM_AUDIO_BLOCKS = AudioEffectPitchShift::ANALYSIS_SIZE / AUD |
|
|
|
constexpr uint32_t FFT_FORWARD = 0; |
|
|
|
constexpr uint32_t FFT_FORWARD = 0; |
|
|
|
constexpr uint32_t FFT_INVERSE = 1; |
|
|
|
constexpr uint32_t FFT_INVERSE = 1; |
|
|
|
constexpr uint32_t FFT_DO_BIT_REVERSE = 1; |
|
|
|
constexpr uint32_t FFT_DO_BIT_REVERSE = 1; |
|
|
|
|
|
|
|
constexpr float WINDOW_GAIN = 0.5; |
|
|
|
|
|
|
|
|
|
|
|
AudioEffectPitchShift::AudioEffectPitchShift() |
|
|
|
AudioEffectPitchShift::AudioEffectPitchShift() |
|
|
|
: AudioStream(1, m_inputQueueArray) |
|
|
|
: AudioStream(1, m_inputQueueArray) |
|
|
|
{ |
|
|
|
{ |
|
|
|
// clear the audio buffer to avoid pops
|
|
|
|
m_analysisBuffer = (fftType_t *)malloc(ANALYSIS_SIZE*sizeof(fftType_t)); |
|
|
|
|
|
|
|
m_windowFunction = (fftType_t *)malloc(ANALYSIS_SIZE*sizeof(fftType_t)); |
|
|
|
|
|
|
|
m_windowBuffer = (fftType_t *)malloc(SYNTHESIS_SIZE*sizeof(fftType_t)); |
|
|
|
|
|
|
|
m_outputBuffer = (fftType_t *)malloc(ANALYSIS_SIZE*sizeof(fftType_t)); |
|
|
|
|
|
|
|
m_synthesisBuffer = (fftType_t *)malloc(SYNTHESIS_SIZE*sizeof(fftType_t)); |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
m_analysisFreqBuffer = (fftType_t *)malloc(2*SYNTHESIS_SIZE*sizeof(fftType_t)); |
|
|
|
|
|
|
|
m_synthesisFreqBuffer = (fftType_t *)malloc(2*SYNTHESIS_SIZE*sizeof(fftType_t)); |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
// clear the audio buffer to avoid pops and configure the Hann window
|
|
|
|
for (unsigned i=0; i<AudioEffectPitchShift::ANALYSIS_SIZE; i++) { |
|
|
|
for (unsigned i=0; i<AudioEffectPitchShift::ANALYSIS_SIZE; i++) { |
|
|
|
m_analysisBuffer[i] = 0.0f; |
|
|
|
m_analysisBuffer[i] = 0.0f; |
|
|
|
|
|
|
|
m_windowFunction[i] = 0.5f * (1.0f - cos(2.0f * M_PI * (float)i / ANALYSIS_SIZE_F)) * WINDOW_GAIN; |
|
|
|
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
for (unsigned i=0; i<SYNTHESIS_SIZE; i++) { |
|
|
|
|
|
|
|
m_windowBuffer[i] = 0.0f; |
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
// Configure the FFT
|
|
|
|
// Configure the FFT
|
|
|
|
// arm_rfft_init_f32(&rfftForwardInst, &cfftForwardInst, AudioEffectPitchShift::ANALYSIS_SIZE,
|
|
|
|
|
|
|
|
// FFT_FORWARD, FFT_DO_BIT_REVERSE);
|
|
|
|
|
|
|
|
// arm_rfft_init_f32(&rfftInverseInst, &cfftInverseInst, AudioEffectPitchShift::SYNTHESIS_SIZE,
|
|
|
|
|
|
|
|
// FFT_INVERSE, FFT_DO_BIT_REVERSE);
|
|
|
|
|
|
|
|
unsigned ret; |
|
|
|
unsigned ret; |
|
|
|
ret = arm_cfft_radix4_init_f32(&cfftForwardInst, ANALYSIS_SIZE, FFT_FORWARD, FFT_DO_BIT_REVERSE); //init FFT
|
|
|
|
ret = arm_rfft_init_f32(&fftFwdReal, &fftFwdComplex, SYNTHESIS_SIZE, FFT_FORWARD, FFT_DO_BIT_REVERSE); //init FFT
|
|
|
|
if (!ret) { m_initFailed = true; }; |
|
|
|
if (ret != ARM_MATH_SUCCESS) { m_initFailed = true; }; |
|
|
|
ret = arm_cfft_radix4_init_f32(&cfftInverseInst, SYNTHESIS_SIZE, FFT_INVERSE, FFT_DO_BIT_REVERSE); //init FFT
|
|
|
|
ret = arm_rfft_init_f32(&fftInvReal, &fftInvComplex, SYNTHESIS_SIZE, FFT_INVERSE, FFT_DO_BIT_REVERSE); //init FFT
|
|
|
|
if (!ret) { m_initFailed = true; }; |
|
|
|
if (ret != ARM_MATH_SUCCESS) { m_initFailed = true; }; |
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
AudioEffectPitchShift::~AudioEffectPitchShift() |
|
|
|
AudioEffectPitchShift::~AudioEffectPitchShift() |
|
|
|
{ |
|
|
|
{ |
|
|
|
|
|
|
|
free(m_analysisBuffer); |
|
|
|
|
|
|
|
m_analysisBuffer = nullptr; |
|
|
|
|
|
|
|
free(m_windowFunction); |
|
|
|
|
|
|
|
m_windowFunction = nullptr; |
|
|
|
|
|
|
|
free(m_windowBuffer); |
|
|
|
|
|
|
|
m_windowBuffer = nullptr; |
|
|
|
|
|
|
|
free(m_outputBuffer); |
|
|
|
|
|
|
|
m_outputBuffer = nullptr; |
|
|
|
|
|
|
|
free(m_synthesisBuffer); |
|
|
|
|
|
|
|
m_synthesisBuffer = nullptr; |
|
|
|
|
|
|
|
free(m_analysisFreqBuffer); |
|
|
|
|
|
|
|
m_analysisFreqBuffer = nullptr; |
|
|
|
|
|
|
|
free(m_synthesisFreqBuffer); |
|
|
|
|
|
|
|
m_synthesisFreqBuffer = nullptr; |
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
void AudioEffectPitchShift::update(void) |
|
|
|
void AudioEffectPitchShift::update(void) |
|
|
@ -71,59 +96,57 @@ void AudioEffectPitchShift::update(void) |
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
// DO PROCESSING HERE
|
|
|
|
// DO PROCESSING HERE
|
|
|
|
// Update the fifo
|
|
|
|
|
|
|
|
// m_inputFifo.push_back(inputAudioBlock); // insert the new block
|
|
|
|
|
|
|
|
// release(m_inputFifo.front()); //
|
|
|
|
|
|
|
|
// m_inputFifo.pop_front();
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
// Convert the contents of the audio blocks to the contiguous buffer
|
|
|
|
// Convert the contents of the audio blocks to the contiguous buffer
|
|
|
|
// 1) Be aware the audio library stores audio samples in reverse temporal order.
|
|
|
|
// 1) Be aware the audio library stores audio samples in reverse temporal order.
|
|
|
|
// This means the first sample (in time) is in the last location of the buffer.
|
|
|
|
// This means the first sample (in time) is in the last location of the buffer.
|
|
|
|
// 2) the oldest audio is at the front of the queue, the latest at the back
|
|
|
|
// 2) the oldest audio is at the front of the queue, the most recent at the back
|
|
|
|
float *analysisPtr = &m_analysisBuffer[0]; |
|
|
|
fftType_t *analysisPtr = &m_analysisBuffer[0]; |
|
|
|
float *analysisFreqPtr = &m_analysisFreqBuffer[0]; |
|
|
|
fftType_t *analysisFreqPtr = &m_analysisFreqBuffer[0]; |
|
|
|
float *synthesisFreqPtr = &m_synthesisFreqBuffer[0]; |
|
|
|
fftType_t *synthesisFreqPtr = &m_synthesisFreqBuffer[0]; |
|
|
|
float *synthesisPtr = &m_synthesisBuffer[0]; |
|
|
|
fftType_t *synthesisPtr = &m_synthesisBuffer[0]; |
|
|
|
|
|
|
|
|
|
|
|
// first shift the contents of the float buffer up by AUDIO_BLOCK SAMPLES
|
|
|
|
// first shift the contents of the fftType_t buffer up by AUDIO_BLOCK SAMPLES
|
|
|
|
for (unsigned i=0; i<NUM_AUDIO_BLOCKS-1; i++) { |
|
|
|
for (unsigned i=0; i<(NUM_AUDIO_BLOCKS-1); i++) { |
|
|
|
memcpy(&analysisPtr[i*AUDIO_BLOCK_SAMPLES], &analysisPtr[(i+1)*AUDIO_BLOCK_SAMPLES], AUDIO_BLOCK_SAMPLES*sizeof(float)); |
|
|
|
memcpy(&analysisPtr[i*AUDIO_BLOCK_SAMPLES], &analysisPtr[(i+1)*AUDIO_BLOCK_SAMPLES], AUDIO_BLOCK_SAMPLES*sizeof(fftType_t)); |
|
|
|
} |
|
|
|
memcpy(&m_outputBuffer[i*AUDIO_BLOCK_SAMPLES], &m_outputBuffer[(i+1)*AUDIO_BLOCK_SAMPLES], AUDIO_BLOCK_SAMPLES*sizeof(fftType_t)); |
|
|
|
// Convert the newest incoming audio block to float
|
|
|
|
} |
|
|
|
arm_q15_to_float(inputAudioBlock->data, &analysisPtr[(NUM_AUDIO_BLOCKS-1)*AUDIO_BLOCK_SAMPLES], AUDIO_BLOCK_SAMPLES); |
|
|
|
// Convert the newest incoming audio block to fftType_t
|
|
|
|
|
|
|
|
int16ToFft(inputAudioBlock->data, &analysisPtr[(NUM_AUDIO_BLOCKS-1)*AUDIO_BLOCK_SAMPLES], AUDIO_BLOCK_SAMPLES); |
|
|
|
|
|
|
|
memset(&m_outputBuffer[(NUM_AUDIO_BLOCKS-1)*AUDIO_BLOCK_SAMPLES], 0, AUDIO_BLOCK_SAMPLES * sizeof(fftType_t)); |
|
|
|
release(inputAudioBlock); // were done with it now
|
|
|
|
release(inputAudioBlock); // were done with it now
|
|
|
|
|
|
|
|
|
|
|
|
//if (m_initFailed) { Serial.println("FFT INIT FAILED"); }
|
|
|
|
if (m_initFailed) { Serial.println("FFT INIT FAILED"); } |
|
|
|
|
|
|
|
|
|
|
|
// Construct the interleaved FFT buffer
|
|
|
|
// Window the contents of the analysis buffer to a temp buffer
|
|
|
|
unsigned idx = 0; |
|
|
|
memset(m_windowBuffer, 0, sizeof(float) * SYNTHESIS_SIZE); |
|
|
|
for (unsigned i=0; i<ANALYSIS_SIZE; i++) { |
|
|
|
arm_mult_f32(analysisPtr, &m_windowFunction[0], m_windowBuffer, ANALYSIS_SIZE); |
|
|
|
m_analysisFreqBuffer[idx] = analysisPtr[i]; |
|
|
|
|
|
|
|
m_analysisFreqBuffer[idx+1] = 0; |
|
|
|
|
|
|
|
idx += 2; |
|
|
|
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
// Perform the FFT
|
|
|
|
// Perform the FFT
|
|
|
|
arm_cfft_radix4_f32(&cfftForwardInst, analysisFreqPtr); |
|
|
|
arm_rfft_f32(&fftFwdReal, m_windowBuffer, analysisFreqPtr); |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
//memset(analysisPtr, 0, sizeof(float)*ANALYSIS_SIZE);
|
|
|
|
|
|
|
|
|
|
|
|
// perform the ocean pitch shift
|
|
|
|
// perform the ocean pitch shift
|
|
|
|
m_ocean(analysisFreqPtr, synthesisFreqPtr, (float)(m_frameIndex), m_pitchScale); |
|
|
|
m_ocean(analysisFreqPtr, synthesisFreqPtr, (float)(m_frameIndex), m_pitchScale); |
|
|
|
//memcpy(synthesisFreqPtr, analysisFreqPtr, 2*ANALYSIS_SIZE*sizeof(float));
|
|
|
|
//memcpy(synthesisFreqPtr, analysisFreqPtr, 2*ANALYSIS_SIZE*sizeof(fftType_t));
|
|
|
|
|
|
|
|
|
|
|
|
// Perform the inverse FFT
|
|
|
|
|
|
|
|
arm_cfft_radix4_f32(&cfftInverseInst, synthesisFreqPtr); |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
// Deinterleave the synthesis buffer
|
|
|
|
arm_rfft_f32(&fftInvReal, synthesisFreqPtr, synthesisPtr); |
|
|
|
idx = 0; |
|
|
|
|
|
|
|
for (unsigned i=0; i<(2*SYNTHESIS_SIZE); i=i+2) { |
|
|
|
// Window the output before adding, only use first part of the synthesized waveform
|
|
|
|
m_synthesisBuffer[idx] = synthesisFreqPtr[i]; |
|
|
|
arm_mult_f32(synthesisPtr, &m_windowFunction[0], synthesisPtr, ANALYSIS_SIZE); |
|
|
|
idx++; |
|
|
|
|
|
|
|
} |
|
|
|
// Add the synthesis to the output buffer
|
|
|
|
|
|
|
|
arm_add_f32(m_outputBuffer, synthesisPtr, m_outputBuffer, ANALYSIS_SIZE); |
|
|
|
|
|
|
|
//memcpy(m_outputBuffer, synthesisPtr, sizeof(fftType_t)*AUDIO_BLOCK_SAMPLES);
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
// Convert the float buffer back to integer
|
|
|
|
// Convert the fftType_t buffer back to integer
|
|
|
|
audio_block_t *outputBlock = allocate(); |
|
|
|
audio_block_t *outputBlock = allocate(); |
|
|
|
arm_float_to_q15 (synthesisPtr, outputBlock->data, AUDIO_BLOCK_SAMPLES); |
|
|
|
//fftToInt16 (analysisPtr, outputBlock->data, AUDIO_BLOCK_SAMPLES);
|
|
|
|
|
|
|
|
fftToInt16 (m_outputBuffer, outputBlock->data, AUDIO_BLOCK_SAMPLES); |
|
|
|
|
|
|
|
|
|
|
|
transmit(outputBlock); |
|
|
|
transmit(outputBlock); |
|
|
|
release(outputBlock); |
|
|
|
release(outputBlock); |
|
|
@ -151,6 +174,15 @@ void AudioEffectPitchShift::processMidi(int channel, int control, int value) |
|
|
|
return; |
|
|
|
return; |
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
if ((m_midiConfig[PITCH][MIDI_CHANNEL] == channel) && |
|
|
|
|
|
|
|
(m_midiConfig[PITCH][MIDI_CONTROL] == control)) { |
|
|
|
|
|
|
|
// Volume
|
|
|
|
|
|
|
|
int pitchCents = roundf((val - 1.0f) * 1200.0f); |
|
|
|
|
|
|
|
Serial.println(String("AudioEffectPitchShift::pitch: ") + pitchCents + String(" cents")); |
|
|
|
|
|
|
|
setPitchShiftCents(pitchCents); |
|
|
|
|
|
|
|
return; |
|
|
|
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
void AudioEffectPitchShift::mapMidiControl(int parameter, int midiCC, int midiChannel) |
|
|
|
void AudioEffectPitchShift::mapMidiControl(int parameter, int midiCC, int midiChannel) |
|
|
@ -162,16 +194,38 @@ void AudioEffectPitchShift::mapMidiControl(int parameter, int midiCC, int midiCh |
|
|
|
m_midiConfig[parameter][MIDI_CONTROL] = midiCC; |
|
|
|
m_midiConfig[parameter][MIDI_CONTROL] = midiCC; |
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
void AudioEffectPitchShift::m_ocean(float *inputFreq, float *outputFreq, float frameIndex, float pitchScale) |
|
|
|
|
|
|
|
|
|
|
|
float AudioEffectPitchShift::setPitchKnob(float val) |
|
|
|
|
|
|
|
{ |
|
|
|
|
|
|
|
int pitchCents = roundf((val - 0.5f)*2.0f * 1200.0f); |
|
|
|
|
|
|
|
float pitchScale = setPitchShiftCents(pitchCents); |
|
|
|
|
|
|
|
return pitchScale; |
|
|
|
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
float AudioEffectPitchShift::setPitchShiftCents(int shiftCents) |
|
|
|
|
|
|
|
{ |
|
|
|
|
|
|
|
constexpr float ROOT_12TH_OF_2 = 1.0594630944; |
|
|
|
|
|
|
|
// alpha = nthroot(2,12)^(pitchShiftCents/100);
|
|
|
|
|
|
|
|
m_pitchScale = powf(ROOT_12TH_OF_2,((float)(shiftCents) / 100.0f)); |
|
|
|
|
|
|
|
return m_pitchScale; |
|
|
|
|
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
void AudioEffectPitchShift::m_ocean(fftType_t *inputFreq, fftType_t *outputFreq, float frameIndex, float pitchScale) |
|
|
|
{ |
|
|
|
{ |
|
|
|
// zero the output buffer
|
|
|
|
// zero the output buffer
|
|
|
|
for (unsigned i=0; i<(2*SYNTHESIS_SIZE); i++) { |
|
|
|
for (unsigned i=0; i<(2*SYNTHESIS_SIZE); i++) { |
|
|
|
outputFreq[i] = 0.0f; |
|
|
|
outputFreq[i] = 0.0f; |
|
|
|
} |
|
|
|
} |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
//pitchScale = 2.0f;
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
// float phaseAdjustFactor = -((2.0f*((float)(M_PI))*frameIndex)
|
|
|
|
|
|
|
|
// / (OVERLAP_FACTOR_F * FFT_OVERSAMPLE_FACTOR_F * SYNTHESIS_SIZE_F));
|
|
|
|
float phaseAdjustFactor = -((2.0f*((float)(M_PI))*frameIndex) |
|
|
|
float phaseAdjustFactor = -((2.0f*((float)(M_PI))*frameIndex) |
|
|
|
/ (OVERLAP_FACTOR_F * FFT_OVERSAMPLE_FACTOR_F * SYNTHESIS_SIZE_F)); |
|
|
|
/ (OVERLAP_FACTOR_F * FFT_OVERSAMPLE_FACTOR_F)); |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
//outputFreq[0] = inputFreq[0];
|
|
|
|
|
|
|
|
//outputFreq[1] = inputFreq[1];
|
|
|
|
for (unsigned k=1; k < SYNTHESIS_SIZE/2; k++) { |
|
|
|
for (unsigned k=1; k < SYNTHESIS_SIZE/2; k++) { |
|
|
|
|
|
|
|
|
|
|
|
float a = (float)k; |
|
|
|
float a = (float)k; |
|
|
@ -181,18 +235,29 @@ void AudioEffectPitchShift::m_ocean(float *inputFreq, float *outputFreq, float f |
|
|
|
float b = std::roundf( (FFT_OVERSAMPLE_FACTOR_F * pitchScale * a)); |
|
|
|
float b = std::roundf( (FFT_OVERSAMPLE_FACTOR_F * pitchScale * a)); |
|
|
|
unsigned b_int = (unsigned)(b); |
|
|
|
unsigned b_int = (unsigned)(b); |
|
|
|
|
|
|
|
|
|
|
|
if (b_int < SYNTHESIS_SIZE/2) { |
|
|
|
//if (b_int <256) {
|
|
|
|
|
|
|
|
if ((b_int < (SYNTHESIS_SIZE/2/2))) { |
|
|
|
|
|
|
|
|
|
|
|
// phaseAdjust = (b-ma) * phaseAdjustFactor
|
|
|
|
// phaseAdjust = (b-ma) * phaseAdjustFactor
|
|
|
|
float phaseAdjust = (b - (FFT_OVERSAMPLE_FACTOR_F * a)) * phaseAdjustFactor; |
|
|
|
float phaseAdjust = (b - (FFT_OVERSAMPLE_FACTOR_F * a)) * phaseAdjustFactor; |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
float a_real = inputFreq[2*k]; |
|
|
|
float a_real = inputFreq[2*k]; |
|
|
|
float a_imag = inputFreq[2*k+1]; |
|
|
|
float a_imag = inputFreq[2*k+1]; |
|
|
|
|
|
|
|
|
|
|
|
outputFreq[2*b_int] = (a_real * arm_cos_f32(phaseAdjust)) - (a_imag * arm_sin_f32(phaseAdjust)); |
|
|
|
// Note the real and imag are interleaved
|
|
|
|
outputFreq[2*b_int+1] = (a_real * arm_sin_f32(phaseAdjust)) + (a_imag * arm_cos_f32(phaseAdjust)); |
|
|
|
unsigned idx = 2*b_int; |
|
|
|
} |
|
|
|
outputFreq[idx] = (a_real * fastCos(phaseAdjust)) - (a_imag * fastSin(phaseAdjust)); |
|
|
|
|
|
|
|
outputFreq[idx+1] = (a_real * fastSin(phaseAdjust)) + (a_imag * fastCos(phaseAdjust)); |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
//if ((int)frameIndex % 512 == 0) {
|
|
|
|
|
|
|
|
//Serial.println(String("b:") + b_int + String(" idx:") + idx + String(" coeff:") + outputFreq[idx] + String(":") + outputFreq[idx+1]); }
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
// Negative Frequencies
|
|
|
|
|
|
|
|
//unsigned negB = SYNTHESIS_SIZE-b_int;
|
|
|
|
|
|
|
|
//outputFreq[2*negB] = outputFreq[idx];
|
|
|
|
|
|
|
|
//outputFreq[2*negB+1] = -outputFreq[idx+1];
|
|
|
|
|
|
|
|
} |
|
|
|
// update the imag components
|
|
|
|
// update the imag components
|
|
|
|
} |
|
|
|
} |
|
|
|
} |
|
|
|
} |
|
|
|