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Added low pass filters in the delay path

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master
Steve Lascos 7 years ago
parent 874903b51c
commit 442f871e01
4 changed files with 249 additions and 50 deletions
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  1. 84
      src/AudioEffectAnalogDelay.cpp
  2. 16
      src/AudioEffectAnalogDelay.h
  3. 139
      src/LibBasicFunctions.cpp
  4. 60
      src/LibBasicFunctions.h

84
src/AudioEffectAnalogDelay.cpp
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@ -13,17 +13,28 @@ constexpr int MIDI_NUM_PARAMS = 4;
constexpr int MIDI_CHANNEL = 0; constexpr int MIDI_CHANNEL = 0;
constexpr int MIDI_CONTROL = 1; constexpr int MIDI_CONTROL = 1;
constexpr int MIDI_ENABLE = 0; constexpr int MIDI_BYPASS = 0;
constexpr int MIDI_DELAY = 1; constexpr int MIDI_DELAY = 1;
constexpr int MIDI_FEEDBACK = 2; constexpr int MIDI_FEEDBACK = 2;
constexpr int MIDI_MIX = 3; constexpr int MIDI_MIX = 3;
// BOSS DM-3 Filters
constexpr unsigned NUM_IIR_STAGES = 4;
constexpr unsigned IIR_COEFF_SHIFT = 2;
constexpr int32_t DEFAULT_COEFFS[5*NUM_IIR_STAGES] = {
536870912, 988616936, 455608573, 834606945, -482959709,
536870912, 1031466345, 498793368, 965834205, -467402235,
536870912, 1105821939, 573646688, 928470657, -448083489,
2339, 5093, 2776, 302068995, 4412722
};
AudioEffectAnalogDelay::AudioEffectAnalogDelay(float maxDelay) AudioEffectAnalogDelay::AudioEffectAnalogDelay(float maxDelay)
: AudioStream(1, m_inputQueueArray) : AudioStream(1, m_inputQueueArray)
{ {
m_memory = new AudioDelay(maxDelay); m_memory = new AudioDelay(maxDelay);
m_maxDelaySamples = calcAudioSamples(maxDelay); m_maxDelaySamples = calcAudioSamples(maxDelay);
m_iir = new IirBiQuadFilterHQ(NUM_IIR_STAGES, reinterpret_cast<const int32_t *>(&DEFAULT_COEFFS), IIR_COEFF_SHIFT);
} }
AudioEffectAnalogDelay::AudioEffectAnalogDelay(size_t numSamples) AudioEffectAnalogDelay::AudioEffectAnalogDelay(size_t numSamples)
@ -31,6 +42,7 @@ AudioEffectAnalogDelay::AudioEffectAnalogDelay(size_t numSamples)
{ {
m_memory = new AudioDelay(numSamples); m_memory = new AudioDelay(numSamples);
m_maxDelaySamples = numSamples; m_maxDelaySamples = numSamples;
m_iir = new IirBiQuadFilterHQ(NUM_IIR_STAGES, reinterpret_cast<const int32_t *>(&DEFAULT_COEFFS), IIR_COEFF_SHIFT);
} }
// requires preallocated memory large enough // requires preallocated memory large enough
@ -40,68 +52,64 @@ AudioEffectAnalogDelay::AudioEffectAnalogDelay(ExtMemSlot *slot)
m_memory = new AudioDelay(slot); m_memory = new AudioDelay(slot);
m_maxDelaySamples = slot->size(); m_maxDelaySamples = slot->size();
m_externalMemory = true; m_externalMemory = true;
m_iir = new IirBiQuadFilterHQ(NUM_IIR_STAGES, reinterpret_cast<const int32_t *>(&DEFAULT_COEFFS), IIR_COEFF_SHIFT);
} }
AudioEffectAnalogDelay::~AudioEffectAnalogDelay() AudioEffectAnalogDelay::~AudioEffectAnalogDelay()
{ {
if (m_memory) delete m_memory; if (m_memory) delete m_memory;
if (m_iir) delete m_iir;
} }
void AudioEffectAnalogDelay::update(void) void AudioEffectAnalogDelay::update(void)
{ {
audio_block_t *inputAudioBlock = receiveReadOnly(); // get the next block of input samples audio_block_t *inputAudioBlock = receiveReadOnly(); // get the next block of input samples
if (!inputAudioBlock) { // Check is block is disabled
// create silence
inputAudioBlock = allocate();
if (!inputAudioBlock) { return; } // failed to allocate
else {
clearAudioBlock(inputAudioBlock);
}
}
if (m_enable == false) { if (m_enable == false) {
// release all held memory resources // do not transmit or process any audio, return as quickly as possible.
transmit(inputAudioBlock); if (inputAudioBlock) release(inputAudioBlock);
release(inputAudioBlock); inputAudioBlock = nullptr;
// release all held memory resources
if (m_previousBlock) { if (m_previousBlock) {
release(m_previousBlock); m_previousBlock = nullptr; release(m_previousBlock); m_previousBlock = nullptr;
} }
if (!m_externalMemory) { if (!m_externalMemory) {
// when using internal memory we have to release all references in the ring buffer
while (m_memory->getRingBuffer()->size() > 0) { while (m_memory->getRingBuffer()->size() > 0) {
audio_block_t *releaseBlock = m_memory->getRingBuffer()->front(); audio_block_t *releaseBlock = m_memory->getRingBuffer()->front();
m_memory->getRingBuffer()->pop_front(); m_memory->getRingBuffer()->pop_front();
if (releaseBlock) release(releaseBlock); if (releaseBlock) release(releaseBlock);
} }
} }
return;
} }
if (m_callCount < 1024) { // Check is block is bypassed, if so either transmit input directly or create silence
if (inputAudioBlock) { if (m_bypass == true) {
transmit(inputAudioBlock, 0); // transmit the input directly
release(inputAudioBlock); if (!inputAudioBlock) {
// create silence
inputAudioBlock = allocate();
if (!inputAudioBlock) { return; } // failed to allocate
else {
clearAudioBlock(inputAudioBlock);
}
} }
m_callCount++; return; transmit(inputAudioBlock, 0);
release(inputAudioBlock);
return;
} }
// Otherwise perform normal processing
m_callCount++;
//Serial.println(String("AudioEffectAnalgDelay::update: ") + m_callCount);
// Preprocessing // Preprocessing
audio_block_t *preProcessed = allocate(); audio_block_t *preProcessed = allocate();
// mix the input with the feedback path in the pre-processing stage
m_preProcessing(preProcessed, inputAudioBlock, m_previousBlock); m_preProcessing(preProcessed, inputAudioBlock, m_previousBlock);
audio_block_t *blockToRelease = m_memory->addBlock(preProcessed); audio_block_t *blockToRelease = m_memory->addBlock(preProcessed);
if (blockToRelease) release(blockToRelease); if (blockToRelease) release(blockToRelease);
// if (inputAudioBlock) {
// transmit(inputAudioBlock, 0);
// release(inputAudioBlock);
// }
// return;
// OUTPUT PROCESSING // OUTPUT PROCESSING
audio_block_t *blockToOutput = nullptr; audio_block_t *blockToOutput = nullptr;
blockToOutput = allocate(); blockToOutput = allocate();
@ -110,7 +118,7 @@ void AudioEffectAnalogDelay::update(void)
if (!blockToOutput) return; // skip this time due to failure if (!blockToOutput) return; // skip this time due to failure
// copy over data // copy over data
m_memory->getSamples(blockToOutput, m_delaySamples); m_memory->getSamples(blockToOutput, m_delaySamples);
// perform the mix // perform the wet/dry mix mix
m_postProcessing(blockToOutput, inputAudioBlock, blockToOutput); m_postProcessing(blockToOutput, inputAudioBlock, blockToOutput);
transmit(blockToOutput); transmit(blockToOutput);
@ -177,11 +185,11 @@ void AudioEffectAnalogDelay::processMidi(int channel, int control, int value)
return; return;
} }
if ((m_midiConfig[MIDI_ENABLE][MIDI_CHANNEL] == channel) && if ((m_midiConfig[MIDI_BYPASS][MIDI_CHANNEL] == channel) &&
(m_midiConfig[MIDI_ENABLE][MIDI_CONTROL] == control)) { (m_midiConfig[MIDI_BYPASS][MIDI_CONTROL] == control)) {
// Enable // Bypass
if (value >= 65) { enable(); Serial.println(String("AudioEffectAnalogDelay::enable: ON") + value); } if (value >= 65) { bypass(false); Serial.println(String("AudioEffectAnalogDelay::not bypassed -> ON") + value); }
else { disable(); Serial.println(String("AudioEffectAnalogDelay::enable: OFF") + value); } else { bypass(true); Serial.println(String("AudioEffectAnalogDelay::bypassed -> OFF") + value); }
return; return;
} }
@ -208,10 +216,10 @@ void AudioEffectAnalogDelay::mapMidiDelay(int control, int channel)
m_midiConfig[MIDI_DELAY][MIDI_CONTROL] = control; m_midiConfig[MIDI_DELAY][MIDI_CONTROL] = control;
} }
void AudioEffectAnalogDelay::mapMidiEnable(int control, int channel) void AudioEffectAnalogDelay::mapMidiBypass(int control, int channel)
{ {
m_midiConfig[MIDI_ENABLE][MIDI_CHANNEL] = channel; m_midiConfig[MIDI_BYPASS][MIDI_CHANNEL] = channel;
m_midiConfig[MIDI_ENABLE][MIDI_CONTROL] = control; m_midiConfig[MIDI_BYPASS][MIDI_CONTROL] = control;
} }
void AudioEffectAnalogDelay::mapMidiFeedback(int control, int channel) void AudioEffectAnalogDelay::mapMidiFeedback(int control, int channel)
@ -239,6 +247,8 @@ void AudioEffectAnalogDelay::m_preProcessing(audio_block_t *out, audio_block_t *
void AudioEffectAnalogDelay::m_postProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet) void AudioEffectAnalogDelay::m_postProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet)
{ {
if ( out && dry && wet) { if ( out && dry && wet) {
// Simulate the LPF IIR nature of the analog systems
m_iir->process(wet->data, wet->data, AUDIO_BLOCK_SAMPLES);
alphaBlend(out, dry, wet, m_mix); alphaBlend(out, dry, wet, m_mix);
} else if (dry) { } else if (dry) {
memcpy(out->data, dry->data, sizeof(int16_t) * AUDIO_BLOCK_SAMPLES); memcpy(out->data, dry->data, sizeof(int16_t) * AUDIO_BLOCK_SAMPLES);

16
src/AudioEffectAnalogDelay.h
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@ -8,8 +8,6 @@
#ifndef SRC_AUDIOEFFECTANALOGDELAY_H_ #ifndef SRC_AUDIOEFFECTANALOGDELAY_H_
#define SRC_AUDIOEFFECTANALOGDELAY_H_ #define SRC_AUDIOEFFECTANALOGDELAY_H_
//#include <vector>
#include <Audio.h> #include <Audio.h>
#include "LibBasicFunctions.h" #include "LibBasicFunctions.h"
@ -17,8 +15,6 @@ namespace BAGuitar {
class AudioEffectAnalogDelay : public AudioStream { class AudioEffectAnalogDelay : public AudioStream {
public: public:
static constexpr int MAX_DELAY_CHANNELS = 8;
AudioEffectAnalogDelay() = delete; AudioEffectAnalogDelay() = delete;
AudioEffectAnalogDelay(float maxDelay); AudioEffectAnalogDelay(float maxDelay);
AudioEffectAnalogDelay(size_t numSamples); AudioEffectAnalogDelay(size_t numSamples);
@ -28,35 +24,35 @@ public:
virtual void update(void); virtual void update(void);
void delay(float milliseconds); void delay(float milliseconds);
void delay(size_t delaySamples); void delay(size_t delaySamples);
void bypass(bool byp) { m_bypass = byp; }
void feedback(float feedback) { m_feedback = feedback; } void feedback(float feedback) { m_feedback = feedback; }
void mix(float mix) { m_mix = mix; } void mix(float mix) { m_mix = mix; }
void enable() { m_enable = true; } void enable() { m_enable = true; }
void disable() { m_enable = false; } void disable() { m_enable = false; }
void processMidi(int channel, int control, int value); void processMidi(int channel, int control, int value);
void mapMidiEnable(int control, int channel = 0); void mapMidiBypass(int control, int channel = 0);
void mapMidiDelay(int control, int channel = 0); void mapMidiDelay(int control, int channel = 0);
void mapMidiFeedback(int control, int channel = 0); void mapMidiFeedback(int control, int channel = 0);
void mapMidiMix(int control, int channel = 0); void mapMidiMix(int control, int channel = 0);
private: private:
audio_block_t *m_inputQueueArray[1]; audio_block_t *m_inputQueueArray[1];
bool m_bypass = true;
bool m_enable = false; bool m_enable = false;
bool m_externalMemory = false; bool m_externalMemory = false;
AudioDelay *m_memory = nullptr; AudioDelay *m_memory = nullptr;
size_t m_maxDelaySamples = 0; size_t m_maxDelaySamples = 0;
audio_block_t *m_previousBlock = nullptr;
IirBiQuadFilterHQ *m_iir = nullptr;
// Controls // Controls
int m_midiConfig[4][2]; int m_midiConfig[4][2];
//int m_midiEnable[2] = {0,16};
size_t m_delaySamples = 0; size_t m_delaySamples = 0;
//int m_midiDelay[2] = {0,20};
float m_feedback = 0.0f; float m_feedback = 0.0f;
//int m_midiFeedback[2] = {0,21};
float m_mix = 0.0f; float m_mix = 0.0f;
//int m_midiMix[2] = {0,22};
audio_block_t *m_previousBlock = nullptr;
void m_preProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet); void m_preProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet);
void m_postProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet); void m_postProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet);

139
src/LibBasicFunctions.cpp
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@ -34,9 +34,21 @@ size_t calcOffset(QueuePosition position)
void alphaBlend(audio_block_t *out, audio_block_t *dry, audio_block_t* wet, float mix) void alphaBlend(audio_block_t *out, audio_block_t *dry, audio_block_t* wet, float mix)
{ {
for (int i=0; i< AUDIO_BLOCK_SAMPLES; i++) { //Non-optimized version for illustrative purposes
out->data[i] = (dry->data[i] * (1 - mix)) + (wet->data[i] * mix); // for (int i=0; i< AUDIO_BLOCK_SAMPLES; i++) {
} // out->data[i] = (dry->data[i] * (1 - mix)) + (wet->data[i] * mix);
// }
// return;
// ARM DSP optimized
int16_t wetBuffer[AUDIO_BLOCK_SAMPLES];
int16_t dryBuffer[AUDIO_BLOCK_SAMPLES];
int16_t scaleFractWet = (int16_t)(mix * 32767.0f);
int16_t scaleFractDry = 32767-scaleFractWet;
arm_scale_q15(dry->data, scaleFractDry, 0, dryBuffer, AUDIO_BLOCK_SAMPLES);
arm_scale_q15(wet->data, scaleFractWet, 0, wetBuffer, AUDIO_BLOCK_SAMPLES);
arm_add_q15(wetBuffer, dryBuffer, out->data, AUDIO_BLOCK_SAMPLES);
} }
void clearAudioBlock(audio_block_t *block) void clearAudioBlock(audio_block_t *block)
@ -45,6 +57,9 @@ void clearAudioBlock(audio_block_t *block)
} }
////////////////////////////////////////////////////
// AudioDelay
////////////////////////////////////////////////////
AudioDelay::AudioDelay(size_t maxSamples) AudioDelay::AudioDelay(size_t maxSamples)
: m_slot(nullptr) : m_slot(nullptr)
{ {
@ -210,5 +225,123 @@ bool AudioDelay::getSamples(audio_block_t *dest, size_t offset, size_t numSample
} }
////////////////////////////////////////////////////
// IirBiQuadFilter
////////////////////////////////////////////////////
IirBiQuadFilter::IirBiQuadFilter(unsigned numStages, const int32_t *coeffs, int coeffShift)
: NUM_STAGES(numStages)
{
m_coeffs = new int32_t[5*numStages];
memcpy(m_coeffs, coeffs, 5*numStages * sizeof(int32_t));
m_state = new int32_t[4*numStages];
arm_biquad_cascade_df1_init_q31(&m_iirCfg, numStages, m_coeffs, m_state, coeffShift);
}
IirBiQuadFilter::~IirBiQuadFilter()
{
if (m_coeffs) delete [] m_coeffs;
if (m_state) delete [] m_state;
}
bool IirBiQuadFilter::process(int16_t *output, int16_t *input, size_t numSamples)
{
if (!output) return false;
if (!input) {
// send zeros
memset(output, 0, numSamples * sizeof(int16_t));
} else {
// create convertion buffers on teh stack
int32_t input32[numSamples];
int32_t output32[numSamples];
for (int i=0; i<numSamples; i++) {
input32[i] = (int32_t)(input[i]);
}
arm_biquad_cascade_df1_fast_q31(&m_iirCfg, input32, output32, numSamples);
for (int i=0; i<numSamples; i++) {
output[i] = (int16_t)(output32[i] & 0xffff);
}
}
return true;
}
// HIGH QUALITY
IirBiQuadFilterHQ::IirBiQuadFilterHQ(unsigned numStages, const int32_t *coeffs, int coeffShift)
: NUM_STAGES(numStages)
{
m_coeffs = new int32_t[5*numStages];
memcpy(m_coeffs, coeffs, 5*numStages * sizeof(int32_t));
m_state = new int64_t[4*numStages];;
arm_biquad_cas_df1_32x64_init_q31(&m_iirCfg, numStages, m_coeffs, m_state, coeffShift);
}
IirBiQuadFilterHQ::~IirBiQuadFilterHQ()
{
if (m_coeffs) delete [] m_coeffs;
if (m_state) delete [] m_state;
}
bool IirBiQuadFilterHQ::process(int16_t *output, int16_t *input, size_t numSamples)
{
if (!output) return false;
if (!input) {
// send zeros
memset(output, 0, numSamples * sizeof(int16_t));
} else {
// create convertion buffers on teh stack
int32_t input32[numSamples];
int32_t output32[numSamples];
for (int i=0; i<numSamples; i++) {
input32[i] = (int32_t)(input[i]);
}
arm_biquad_cas_df1_32x64_q31(&m_iirCfg, input32, output32, numSamples);
for (int i=0; i<numSamples; i++) {
output[i] = (int16_t)(output32[i] & 0xffff);
}
}
return true;
}
// FLOAT
IirBiQuadFilterFloat::IirBiQuadFilterFloat(unsigned numStages, const float *coeffs)
: NUM_STAGES(numStages)
{
m_coeffs = new float[5*numStages];
memcpy(m_coeffs, coeffs, 5*numStages * sizeof(float));
m_state = new float[4*numStages];;
arm_biquad_cascade_df2T_init_f32(&m_iirCfg, numStages, m_coeffs, m_state);
}
IirBiQuadFilterFloat::~IirBiQuadFilterFloat()
{
if (m_coeffs) delete [] m_coeffs;
if (m_state) delete [] m_state;
}
bool IirBiQuadFilterFloat::process(float *output, float *input, size_t numSamples)
{
if (!output) return false;
if (!input) {
// send zeros
memset(output, 0, numSamples * sizeof(float));
} else {
arm_biquad_cascade_df2T_f32(&m_iirCfg, input, output, numSamples);
}
return true;
}
} }

60
src/LibBasicFunctions.h
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@ -23,6 +23,7 @@
#include <cstddef> #include <cstddef>
#include <new> #include <new>
#include <arm_math.h>
#include "Arduino.h" #include "Arduino.h"
#include "Audio.h" #include "Audio.h"
@ -153,6 +154,65 @@ private:
ExtMemSlot *m_slot = nullptr; ///< When using EXTERNAL memory, an ExtMemSlot must be provided. ExtMemSlot *m_slot = nullptr; ///< When using EXTERNAL memory, an ExtMemSlot must be provided.
}; };
/**************************************************************************//**
* IIR BiQuad Filter - Direct Form I <br>
* y[n] = b0 * x[n] + b1 * x[n-1] + b2 * x[n-2] + a1 * y[n-1] + a2 * y[n-2]<br>
* Some design tools (like Matlab assume the feedback coefficients 'a' are negated. You
* may have to negate your 'a' coefficients.
* @details Note that the ARM CMSIS-DSP library requires an extra zero between first
* and second 'b' coefficients. E.g. <br>
* {b10, 0, b11, b12, a11, a12, b20, 0, b21, b22, a21, a22, ...}
*****************************************************************************/
class IirBiQuadFilter {
public:
IirBiQuadFilter() = delete;
IirBiQuadFilter(unsigned numStages, const int32_t *coeffs, int coeffShift = 0);
virtual ~IirBiQuadFilter();
bool process(int16_t *output, int16_t *input, size_t numSamples);
private:
const unsigned NUM_STAGES;
int32_t *m_coeffs = nullptr;
// ARM DSP Math library filter instance
arm_biquad_casd_df1_inst_q31 m_iirCfg;
int32_t *m_state = nullptr;
};
class IirBiQuadFilterHQ {
public:
IirBiQuadFilterHQ() = delete;
IirBiQuadFilterHQ(unsigned numStages, const int32_t *coeffs, int coeffShift = 0);
virtual ~IirBiQuadFilterHQ();
bool process(int16_t *output, int16_t *input, size_t numSamples);
private:
const unsigned NUM_STAGES;
int32_t *m_coeffs = nullptr;
// ARM DSP Math library filter instance
arm_biquad_cas_df1_32x64_ins_q31 m_iirCfg;
int64_t *m_state = nullptr;
};
class IirBiQuadFilterFloat {
public:
IirBiQuadFilterFloat() = delete;
IirBiQuadFilterFloat(unsigned numStages, const float *coeffs);
virtual ~IirBiQuadFilterFloat();
bool process(float *output, float *input, size_t numSamples);
private:
const unsigned NUM_STAGES;
float *m_coeffs = nullptr;
// ARM DSP Math library filter instance
arm_biquad_cascade_df2T_instance_f32 m_iirCfg;
float *m_state = nullptr;
};
/**************************************************************************//** /**************************************************************************//**
* Customer RingBuffer with random access * Customer RingBuffer with random access
*****************************************************************************/ *****************************************************************************/

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