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converted to heap arrays to save stack space

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feature/pitchShifter
Steve Lascos 5 years ago
parent 070c305322
commit e67043595f
3 changed files with 175 additions and 70 deletions
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  1. 20
      examples/Modulation/PitchShiftExpansion/PitchShiftExpansion.ino
  2. 60
      src/AudioEffectPitchShift.h
  3. 165
      src/effects/AudioEffectPitchShift.cpp

20
examples/Modulation/PitchShiftExpansion/PitchShiftExpansion.ino
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@ -45,7 +45,7 @@ AudioConnection rightOut(cabFilter,0, i2sOut, 1);
//////////////////////////////////////////
// SETUP PHYSICAL CONTROLS
// - POT1 (left) will control the rate
// - POT1 (left) will control the pitch
// - POT2 (right) will control the depth
// - POT3 (centre) will control the volume
// - SW1 (left) will be used as a bypass control
@ -68,7 +68,7 @@ constexpr unsigned MAX_HEADPHONE_VOL = 10;
unsigned headphoneVolume = 8; // control headphone volume from 0 to 10.
// BAPhysicalControls returns a handle when you register a new control. We'll uses these handles when working with the controls.
int bypassHandle, volumeHandle, led1Handle, led2Handle; // Handles for the various controls
int bypassHandle, volumeHandle, pitchHandle, led1Handle, led2Handle; // Handles for the various controls
void setup() {
delay(100); // wait a bit for serial to be available
@ -80,7 +80,7 @@ void setup() {
bypassHandle = controls.addSwitch(BA_EXPAND_SW1_PIN); // will be used for bypass control
//button2Handle = controls.addSwitch(BA_EXPAND_SW2_PIN); // will be used for stepping through filters
// pots
//rateHandle = controls.addPot(BA_EXPAND_POT1_PIN, potCalibMin, potCalibMax, potSwapDirection); // control the amount of delay
pitchHandle = controls.addPot(BA_EXPAND_POT1_PIN, potCalibMin, potCalibMax, potSwapDirection); // control the amount of delay
//depthHandle = controls.addPot(BA_EXPAND_POT2_PIN, potCalibMin, potCalibMax, potSwapDirection);
volumeHandle = controls.addPot(BA_EXPAND_POT3_PIN, potCalibMin, potCalibMax, potSwapDirection);
// leds
@ -102,6 +102,7 @@ void setup() {
// Set some default values.
// These can be changed using the controls on the Blackaddr Audio Expansion Board
pitchShift.setPitchShiftCents(0);
pitchShift.bypass(false);
// Guitar cabinet: Setup 2-stages of LPF, cutoff 4500 Hz, Q-factor 0.7071 (a 'normal' Q-factor)
@ -127,12 +128,13 @@ void loop() {
// if (controls.isSwitchToggled(waveformHandle)) {
// }
// // Use POT1 (left) to control the rate setting
// if (controls.checkPotValue(rateHandle, potValue)) {
// // Pot has changed
// Serial.println(String("New RATE setting: ") + potValue);
// pitchShift.rate(potValue);
// }
// Use POT1 (left) to control the rate setting
if (controls.checkPotValue(pitchHandle, potValue)) {
// Pot has changed
float pitch = pitchShift.setPitchKnob(potValue);
Serial.println(String("New PITCH setting: ") + potValue + String(" / ") + pitch);
}
// // Use POT2 (right) to control the depth setting
// if (controls.checkPotValue(depthHandle, potValue)) {

60
src/AudioEffectPitchShift.h
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@ -26,6 +26,22 @@
#include "BATypes.h"
#include "LibBasicFunctions.h"
//#define USE_INT32
#define USE_FLOAT
//#define USE_INT16
#ifdef USE_FLOAT
using fftType_t = float;
using fftInstance_t = arm_cfft_radix4_instance_f32;
#define fftInit(A,B,C,D) arm_cfft_radix4_init_f32(A,B,C,D);
#define int16ToFft(A,B,C) arm_q15_to_float(A,B,C)
#define fftToInt16(A,B,C) arm_float_to_q15(A,B,C)
#define fft(A,B) arm_cfft_radix4_f32(A,B)
#define fastCos(X) arm_cos_f32(X)
#define fastSin(X) arm_sin_f32(X)
#endif
namespace BAEffects {
/**************************************************************************//**
@ -35,16 +51,21 @@ class AudioEffectPitchShift : public AudioStream {
public:
static constexpr unsigned ANALYSIS_SIZE = 1024;
static constexpr float ANALYSIS_SIZE_F = (float)ANALYSIS_SIZE;
static constexpr unsigned FFT_OVERSAMPLE_FACTOR = 1;
static constexpr float FFT_OVERSAMPLE_FACTOR_F = 1.0f;
static constexpr float FFT_OVERSAMPLE_FACTOR_F = (float)(FFT_OVERSAMPLE_FACTOR);
static constexpr unsigned SYNTHESIS_SIZE = ANALYSIS_SIZE * FFT_OVERSAMPLE_FACTOR;
static constexpr float SYNTHESIS_SIZE_F = (float)(ANALYSIS_SIZE * FFT_OVERSAMPLE_FACTOR);
static constexpr float SYNTHESIS_SIZE_F = (float)(SYNTHESIS_SIZE);
static constexpr float OVERLAP_FACTOR_F = (float)ANALYSIS_SIZE / (float)AUDIO_BLOCK_SAMPLES;
///< List of AudioEffectTremolo MIDI controllable parameters
enum {
BYPASS = 0, ///< controls effect bypass
VOLUME, ///< controls the output volume level
PITCH, ///< controls the pitch scaling factor
NUM_CONTROLS ///< this can be used as an alias for the number of MIDI controls
};
@ -72,6 +93,11 @@ public:
/// @param vol Sets the output volume between -1.0 and +1.0
void volume(float vol) {m_volume = vol; }
float setPitchKnob(float val);
float setPitchShiftCents(int shiftCents);
//void pitch(float pitchScale) { m_pitchScale = pitchScale; }
// ** ENABLE / DISABLE **
/// Enables audio processing. Note: when not enabled, CPU load is nearly zero.
@ -107,18 +133,28 @@ public:
private:
audio_block_t *m_inputQueueArray[1];
//BALibrary::RingBuffer<audio_block_t*> m_inputFifo = BALibrary::RingBuffer<audio_block_t*>(ANALYSIS_SIZE/AUDIO_BLOCK_SAMPLES);
float m_analysisBuffer[ANALYSIS_SIZE];
float m_analysisFreqBuffer[2*ANALYSIS_SIZE];
float m_synthesisFreqBuffer[2*SYNTHESIS_SIZE];
float m_synthesisBuffer[SYNTHESIS_SIZE];
//fftType_t m_analysisBuffer[ANALYSIS_SIZE];
//float m_windowFunction[ANALYSIS_SIZE];
//fftType_t m_analysisFreqBuffer[2*SYNTHESIS_SIZE];
//fftType_t m_synthesisFreqBuffer[2*SYNTHESIS_SIZE];
//fftType_t m_synthesisBuffer[SYNTHESIS_SIZE];
fftType_t *m_analysisBuffer = nullptr;
fftType_t *m_windowFunction = nullptr;
fftType_t *m_analysisFreqBuffer = nullptr;
fftType_t *m_synthesisFreqBuffer = nullptr;
fftType_t *m_synthesisBuffer = nullptr;
fftType_t *m_windowBuffer = nullptr;
fftType_t *m_outputBuffer = nullptr;
bool m_initFailed = false;
unsigned m_frameIndex = 0;
// arm_rfft_instance_f32 fftFwdReal, fftInvReal;
// arm_cfft_radix4_instance_f32 fftFwdComplex, fftInvComplex;
arm_rfft_instance_f32 fftFwdReal, fftInvReal;
arm_cfft_radix4_instance_f32 fftFwdComplex, fftInvComplex;
// float32_t *bufInputReal;
// float32_t *bufInputComplex;
// float32_t *bufOutputReal;
@ -126,7 +162,7 @@ private:
//arm_cfft_radix4_instance_f32 fft_inst_fwd, fft_inst_inv;
//arm_rfft_instance_f32 rfftForwardInst, rfftInverseInst;
arm_cfft_radix4_instance_f32 cfftForwardInst, cfftInverseInst;
fftInstance_t cfftForwardInst, cfftInverseInst;
//uint8_t ifftFlag = 0; // 0 is FFT, 1 is IFFT
//uint8_t doBitReverse = 1;
@ -138,8 +174,10 @@ private:
float m_volume = 1.0f;
float m_pitchScale = 1.0f;
int m_shiftCents = 0;
void m_ocean(float *inputFreq, float *outputFreq, float frameIndex, float pitchScale);
void m_ocean(fftType_t *inputFreq, fftType_t *outputFreq, float frameIndex, float pitchScale);
//void m_ocean16(int16_t* inputFreq, int16_t* outputFreq, float frameIndex, float pitchScale);
};

165
src/effects/AudioEffectPitchShift.cpp
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@ -18,29 +18,54 @@ constexpr unsigned NUM_AUDIO_BLOCKS = AudioEffectPitchShift::ANALYSIS_SIZE / AUD
constexpr uint32_t FFT_FORWARD = 0;
constexpr uint32_t FFT_INVERSE = 1;
constexpr uint32_t FFT_DO_BIT_REVERSE = 1;
constexpr float WINDOW_GAIN = 0.5;
AudioEffectPitchShift::AudioEffectPitchShift()
: 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++) {
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
// 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;
ret = arm_cfft_radix4_init_f32(&cfftForwardInst, ANALYSIS_SIZE, FFT_FORWARD, FFT_DO_BIT_REVERSE); //init FFT
if (!ret) { m_initFailed = true; };
ret = arm_cfft_radix4_init_f32(&cfftInverseInst, SYNTHESIS_SIZE, FFT_INVERSE, FFT_DO_BIT_REVERSE); //init FFT
if (!ret) { m_initFailed = true; };
ret = arm_rfft_init_f32(&fftFwdReal, &fftFwdComplex, SYNTHESIS_SIZE, FFT_FORWARD, FFT_DO_BIT_REVERSE); //init FFT
if (ret != ARM_MATH_SUCCESS) { m_initFailed = true; };
ret = arm_rfft_init_f32(&fftInvReal, &fftInvComplex, SYNTHESIS_SIZE, FFT_INVERSE, FFT_DO_BIT_REVERSE); //init FFT
if (ret != ARM_MATH_SUCCESS) { m_initFailed = true; };
}
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)
@ -71,59 +96,57 @@ void AudioEffectPitchShift::update(void)
}
// 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
// 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.
// 2) the oldest audio is at the front of the queue, the latest at the back
float *analysisPtr = &m_analysisBuffer[0];
float *analysisFreqPtr = &m_analysisFreqBuffer[0];
float *synthesisFreqPtr = &m_synthesisFreqBuffer[0];
float *synthesisPtr = &m_synthesisBuffer[0];
// first shift the contents of the float buffer up by AUDIO_BLOCK SAMPLES
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));
// 2) the oldest audio is at the front of the queue, the most recent at the back
fftType_t *analysisPtr = &m_analysisBuffer[0];
fftType_t *analysisFreqPtr = &m_analysisFreqBuffer[0];
fftType_t *synthesisFreqPtr = &m_synthesisFreqBuffer[0];
fftType_t *synthesisPtr = &m_synthesisBuffer[0];
// first shift the contents of the fftType_t buffer up by AUDIO_BLOCK SAMPLES
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(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
//if (m_initFailed) { Serial.println("FFT INIT FAILED"); }
if (m_initFailed) { Serial.println("FFT INIT FAILED"); }
// Construct the interleaved FFT buffer
unsigned idx = 0;
for (unsigned i=0; i<ANALYSIS_SIZE; i++) {
m_analysisFreqBuffer[idx] = analysisPtr[i];
m_analysisFreqBuffer[idx+1] = 0;
idx += 2;
}
// Window the contents of the analysis buffer to a temp buffer
memset(m_windowBuffer, 0, sizeof(float) * SYNTHESIS_SIZE);
arm_mult_f32(analysisPtr, &m_windowFunction[0], m_windowBuffer, ANALYSIS_SIZE);
// 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
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
idx = 0;
for (unsigned i=0; i<(2*SYNTHESIS_SIZE); i=i+2) {
m_synthesisBuffer[idx] = synthesisFreqPtr[i];
idx++;
}
arm_rfft_f32(&fftInvReal, synthesisFreqPtr, synthesisPtr);
// Convert the float buffer back to integer
// Window the output before adding, only use first part of the synthesized waveform
arm_mult_f32(synthesisPtr, &m_windowFunction[0], synthesisPtr, ANALYSIS_SIZE);
// 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 fftType_t buffer back to integer
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);
release(outputBlock);
@ -151,6 +174,15 @@ void AudioEffectPitchShift::processMidi(int channel, int control, int value)
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)
@ -162,16 +194,38 @@ void AudioEffectPitchShift::mapMidiControl(int parameter, int midiCC, int midiCh
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
for (unsigned i=0; i<(2*SYNTHESIS_SIZE); i++) {
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)
/ (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++) {
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));
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
float phaseAdjust = (b - (FFT_OVERSAMPLE_FACTOR_F * a)) * phaseAdjustFactor;
float a_real = inputFreq[2*k];
float a_imag = inputFreq[2*k+1];
outputFreq[2*b_int] = (a_real * arm_cos_f32(phaseAdjust)) - (a_imag * arm_sin_f32(phaseAdjust));
outputFreq[2*b_int+1] = (a_real * arm_sin_f32(phaseAdjust)) + (a_imag * arm_cos_f32(phaseAdjust));
}
// Note the real and imag are interleaved
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
}
}

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