Basic working chorus, still needs range tweaking, LFO filtering, etc.

examples/Modulation/AnalogChorusDemoExpansion/AnalogChorusDemoExpansion.ino

@@ -87,7 +87,7 @@ int bypassHandle, filterHandle, rateHandle, depthHandle, mixHandle, led1Handle,
 void setup() {
   delay(100); // wait a bit for serial to be available
   Serial.begin(57600); // Start the serial port
-  delay(100);
+  delay(500);
 
   // Setup the controls. The return value is the handle to use when checking for control changes, etc.
   // pushbuttons
@@ -137,6 +137,7 @@ void setup() {
   //analogChorus.setFilter(AudioEffectAnalogChorus::Filter::CE2); // The default filter. Naturally bright echo (highs stay, lows fade away)
   //analogChorus.setFilter(AudioEffectAnalogChorus::Filter::WARM); // A warm filter with a smooth frequency rolloff above 2Khz
   //analogChorus.setFilter(AudioEffectAnalogChorus::Filter::DARK); // A very dark filter, with a sharp rolloff above 1Khz
+  analogChorus.setFilter(AudioEffectAnalogChorus::Filter::WARM); // A very dark filter, with a sharp rolloff above 1Khz
 
   // Guitar cabinet: Setup 2-stages of LPF, cutoff 4500 Hz, Q-factor 0.7071 (a 'normal' Q-factor)
   cabFilter.setLowpass(0, 4500, .7071);
@@ -157,30 +158,39 @@ void loop() {
   }
 
   // Use SW2 to cycle through the filters
+//  controls.setOutput(led2Handle, controls.getSwitchValue(led2Handle));
+//  if (controls.isSwitchToggled(filterHandle)) {
+//    filterIndex = (filterIndex + 1) % 3; // update and potentionall roll the counter 0, 1, 2, 0, 1, 2, ...
+//    // cast the index between 0 to 2 to the enum class AudioEffectAnalogChorus::Filter
+//    analogChorus.setFilter(static_cast<AudioEffectAnalogChorus::Filter>(filterIndex)); // will cycle through 0 to 2
+//    Serial.println(String("Filter set to ") + filterIndex);
+//  }
+
+  // Use SW2 to cycle through the waveforms
   controls.setOutput(led2Handle, controls.getSwitchValue(led2Handle));
   if (controls.isSwitchToggled(filterHandle)) {
-    filterIndex = (filterIndex + 1) % 3; // update and potentionall roll the counter 0, 1, 2, 0, 1, 2, ...
+    filterIndex = (filterIndex + 1) % 4; // update and potentionall roll the counter 0, 1, 2, 0, 1, 2, ...
-    // cast the index between 0 to 2 to the enum class AudioEffectAnalogChorus::Filter
+    // cast the index between 0 to 3 to the enum class AudioEffectAnalogChorus::Filter
-    analogChorus.setFilter(static_cast<AudioEffectAnalogChorus::Filter>(filterIndex)); // will cycle through 0 to 2
+    analogChorus.setWaveform(static_cast<Waveform>(filterIndex)); // will cycle through 0 to 2
-    Serial.println(String("Filter set to ") + filterIndex);
+    Serial.println(String("Waveform set to ") + filterIndex);
   }
 
   // Use POT1 (left) to control the rate setting
-  if (controls.checkPotValue(rateHandle, potValue)) {
+  if ( (controls.checkPotValue(rateHandle, potValue)) || (loopCount == 0) ) {
     // Pot has changed
     Serial.println(String("New RATE setting: ") + potValue);
     analogChorus.rate(potValue);
   }
 
   // Use POT2 (right) to control the depth setting
-  if (controls.checkPotValue(depthHandle, potValue)) {
+  if ( (controls.checkPotValue(depthHandle, potValue)) || (loopCount == 0) ) {
     // Pot has changed
     Serial.println(String("New DEPTH setting: ") + potValue);
     analogChorus.depth(potValue);
   }
 
   // Use POT3 (centre) to control the mix setting
-  if (controls.checkPotValue(mixHandle, potValue)) {
+  if ( (controls.checkPotValue(mixHandle, potValue)) || (loopCount == 0) ) {
     // Pot has changed
     Serial.println(String("New MIX setting: ") + potValue);
     analogChorus.mix(potValue);

src/AudioEffectAnalogChorus.h

@@ -90,6 +90,10 @@ public:
 	/// Toggle the bypass effect
 	void toggleBypass() { m_bypass = !m_bypass; }
 
+	/// Set the LFO waveform
+	/// @param waveform the LFO waveform to modulate chorus delay with
+	void setWaveform(BALibrary::Waveform waveform);
+
     /// Set the LFO frequency where 0.0f is MIN and 1.0f is MAX
     void rate(float rate);
 
@@ -160,7 +164,7 @@ public:
 	virtual void update(void); ///< update automatically called by the Teesny Audio Library
 
 private:
-	static constexpr float m_DEFAULT_DELAY_MS = 20.0f; ///< default average delay of chorus in milliseconds
+	static constexpr float m_DEFAULT_AVERAGE_DELAY_MS = 20.0f; ///< default average delay of chorus in milliseconds
 	static constexpr float m_DELAY_RANGE      = 15.0f; ///< default range of delay variation in milliseconds
 	static constexpr float m_LFO_MIN_RATE     = 2.0f;  ///< slowest possible LFO rate in milliseconds
 	static constexpr float m_LFO_RANGE        = 8.0f;  ///< fastest possible LFO rate in milliseconds
@@ -171,20 +175,24 @@ private:
 	bool m_enable = false;
 	bool m_externalMemory = false;
 	BALibrary::AudioDelay *m_memory = nullptr;
-	BALibrary::LowFrequencyOscillatorVector<float> m_lfo;
-	size_t m_maxDelaySamples = 0;
     audio_block_t *m_previousBlock = nullptr;
     audio_block_t *m_blockToRelease  = nullptr;
+
+	BALibrary::LowFrequencyOscillatorVector<float> m_lfo;
+	size_t m_maxDelaySamples = 0;
+	//size_t m_currentDelayOffset = 0;
+	float m_delayRange = 0;
+
 	BALibrary::IirBiQuadFilterHQ *m_iir = nullptr;
 
 	// Controls
 	int m_midiConfig[NUM_CONTROLS][2]; // stores the midi parameter mapping
-	size_t m_delaySamples = 0;
+	float m_averageDelaySamples = 0;
 	float m_lfoDepth = 0.0f;
 	float m_mix = 0.0f;
 	float m_volume = 1.0f;
 
-	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);
 
 	// Coefficients

src/LibBasicFunctions.h

@@ -191,6 +191,20 @@ public:
     /// @returns true on success, false on error.
     bool interpolateDelay(int16_t *extendedSourceBuffer, int16_t *destBuffer, float fraction, size_t numSamples = AUDIO_BLOCK_SAMPLES);
 
+    /// Provides linearly interpolated samples between discrete samples in the sample buffer. The interpolation point for each samples
+    /// comes from a provided vector of floats which values between 0.0f and 1.0f;
+    /// The SOURCE buffer MUST BE OVERSIZED
+    /// to numSamples+1. This is because the last output sample is interpolated from between NUM_SAMPLES and NUM_SAMPLES+1.
+    /// @details this function is typically not used with audio blocks directly since you need AUDIO_BLOCK_SAMPLES+1 as the source size
+    /// even though output size is still only AUDIO_BLOCK_SAMPLES. Manually create an oversized buffer and fill it with AUDIO_BLOCK_SAMPLES+1.
+    /// e.g. 129 instead of 128 samples. The destBuffer does not need to be oversized.
+    /// @param extendedSourceBuffer A source array that contains one more input sample than output samples needed.
+    /// @param dest pointer to the target sample array to write the samples to.
+    /// @param fraction a vector of values between 0.0f and 1.0f that sets the interpolation point between the discrete samples.
+    /// @param numSamples number of samples to transfer
+    /// @returns true on success, false on error.
+    bool interpolateDelayVector(int16_t *extendedSourceBuffer, int16_t *destBuffer, float *fractionVector, size_t numSamples = AUDIO_BLOCK_SAMPLES);
+
     /// When using EXTERNAL memory, this function can return a pointer to the underlying ExtMemSlot object associated
     /// with the buffer.
     /// @returns pointer to the underlying ExtMemSlot.

src/common/AudioDelay.cpp

@@ -209,6 +209,22 @@ bool AudioDelay::interpolateDelay(int16_t *extendedSourceBuffer, int16_t *destBu
 	return true;
 }
 
+bool AudioDelay::interpolateDelayVector(int16_t *extendedSourceBuffer, int16_t *destBuffer, float *fractionVector, size_t numSamples)
+{
+    int16_t frac1Vec[numSamples];
+    int16_t frac2Vec[numSamples];
+
+    for (int i=0; i<numSamples; i++) {
+        int16_t fracAsInt = static_cast<int16_t>(32767.0f * fractionVector[i]);
+        frac1Vec[i] = fracAsInt;
+        frac2Vec[i] = 32767-frac1Vec[i];
+
+        destBuffer[i] =  (( frac1Vec[i] * extendedSourceBuffer[i]) >> 16) +  ((frac2Vec[i] * extendedSourceBuffer[i+1]) >> 16);
+    }
+
+    return true;
+}
+
 
 }
 

src/effects/AudioEffectAnalogChorus.cpp

@@ -5,24 +5,30 @@
  *      Author: slascos
  */
 #include <new>
+#include <cmath>
 #include "AudioEffectAnalogChorusFilters.h"
 #include "AudioEffectAnalogChorus.h"
 
 using namespace BALibrary;
 
-//#define INTERPOLATED_DELAY Uncomment this line to test the inteprolated delay which adds 1/10th of a sample
-
 namespace BAEffects {
 
 constexpr int MIDI_CHANNEL = 0;
 constexpr int MIDI_CONTROL = 1;
 
+constexpr float DELAY_REFERENCE_F = static_cast<float>(AUDIO_BLOCK_SAMPLES/2);
+
 AudioEffectAnalogChorus::AudioEffectAnalogChorus()
 : AudioStream(1, m_inputQueueArray)
 {
-    m_memory = new AudioDelay(m_DEFAULT_DELAY_MS + m_DELAY_RANGE);
+    m_memory = new AudioDelay(m_DEFAULT_AVERAGE_DELAY_MS + m_DELAY_RANGE);
-    m_maxDelaySamples = calcAudioSamples(m_DEFAULT_DELAY_MS + m_DELAY_RANGE);
+    m_maxDelaySamples = calcAudioSamples(m_DEFAULT_AVERAGE_DELAY_MS + m_DELAY_RANGE);
+    m_averageDelaySamples = static_cast<float>(calcAudioSamples(m_DEFAULT_AVERAGE_DELAY_MS));
+    m_delayRange          = static_cast<float>(calcAudioSamples(m_DELAY_RANGE));
+
     m_constructFilter();
+    m_lfo.setWaveform(Waveform::TRIANGLE);
+    m_lfo.setRateAudio(4.0f); // Default to 4 Hz
 }
 
 // requires preallocated memory large enough
@@ -31,8 +37,13 @@ AudioEffectAnalogChorus::AudioEffectAnalogChorus(ExtMemSlot *slot)
 {
 	m_memory = new AudioDelay(slot);
 	m_maxDelaySamples = (slot->size() / sizeof(int16_t));
+	m_averageDelaySamples = static_cast<float>(calcAudioSamples(m_DEFAULT_AVERAGE_DELAY_MS));
+	m_delayRange          = static_cast<float>(calcAudioSamples(m_DELAY_RANGE));
+
 	m_externalMemory = true;
 	m_constructFilter();
+	m_lfo.setWaveform(Waveform::TRIANGLE);
+	m_lfo.setRateAudio(4.0f); // Default to 4 Hz
 }
 
 AudioEffectAnalogChorus::~AudioEffectAnalogChorus()
@@ -48,6 +59,19 @@ void AudioEffectAnalogChorus::m_constructFilter(void)
 	m_iir = new IirBiQuadFilterHQ(CE2_NUM_STAGES, reinterpret_cast<const int32_t *>(&CE2), CE2_COEFF_SHIFT);
 }
 
+void AudioEffectAnalogChorus::setWaveform(BALibrary::Waveform waveform)
+{
+    switch(waveform) {
+    case Waveform::SINE :
+    case Waveform::TRIANGLE :
+    case Waveform::SAWTOOTH :
+        m_lfo.setWaveform(waveform);
+        break;
+    default :
+        Serial.println("AudioEffectAnalogChorus::setWaveform: Unsupported Waveform");
+    }
+}
+
 void AudioEffectAnalogChorus::setFilterCoeffs(int numStages, const int32_t *coeffs, int coeffShift)
 {
 	m_iir->changeFilterCoeffs(numStages, coeffs, coeffShift);
@@ -75,7 +99,7 @@ void AudioEffectAnalogChorus::update(void)
 
 	// Check is block is disabled
 	if (m_enable == false) {
-		// do not transmit or process any audio, return as quickly as possible.
+		// do not transmit or proess any audio, return as quickly as possible.
 		if (inputAudioBlock) release(inputAudioBlock);
 
 		// release all held memory resources
@@ -118,14 +142,29 @@ void AudioEffectAnalogChorus::update(void)
 
     // get the data. If using external memory with DMA, this won't be filled until
     // later.
-#ifdef INTERPOLATED_DELAY
+    // We need to grab two blocks of audio since the modulating delay value from the LFO
-    int16_t extendedBuffer[AUDIO_BLOCK_SAMPLES+1]; // need one more sample for intepolating between 128th and 129th (last sample)
+    // can exceed the length of one audio block during the time frame of one audio block.
-    m_memory->getSamples(extendedBuffer, m_delaySamples, AUDIO_BLOCK_SAMPLES+1);
+    int16_t extendedBuffer[(2*AUDIO_BLOCK_SAMPLES)]; // need one more sample for interpolating between 128th and 129th (last sample)
-#else
+
-    m_memory->getSamples(blockToOutput, m_delaySamples);
+    // Get next vector of lfo values, they will range range from -1.0 to +1.0f.
-#endif
+    float *lfoValues = m_lfo.getNextVector();
-
+    //float lfoValues[128];
-
+    for (int i=0; i<128; i++) { lfoValues[i] = lfoValues[i] * m_lfoDepth; }
+
+    // Calculate the starting delay from the first lfo sample. This will represent the 'reference' delay
+    // for this output block
+    float referenceDelay  = m_averageDelaySamples + (lfoValues[0] * m_delayRange);
+    unsigned delaySamples = static_cast<unsigned>(referenceDelay); // round down to the nearest audio sample for indexing into AudioDelay class
+
+    // From a given current delay value, while reading out the next 128, the delay could slew up or down
+    // AUDIO_BLOCK_SAMPLES/2 cycles of delay. For example...
+    // Pitching up  : current + 128 +  64
+    // Pitching down: current -  64 + 128
+    // We need to grab 2*AUDIO_BLOCK_SAMPLES. Be aware that audio samples are stored BACKWARDS in the buffers.
+//    m_memory->getSamples(extendedBuffer                      , delaySamples - (AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
+//    m_memory->getSamples(extendedBuffer + AUDIO_BLOCK_SAMPLES, delaySamples +( AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
+    m_memory->getSamples(extendedBuffer + AUDIO_BLOCK_SAMPLES, delaySamples - (AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
+    m_memory->getSamples(extendedBuffer                      , delaySamples +( AUDIO_BLOCK_SAMPLES/2), AUDIO_BLOCK_SAMPLES);
     // If using DMA, we need something else to do while that read executes, so
     // move on to input preprocessing
 
@@ -142,16 +181,39 @@ void AudioEffectAnalogChorus::update(void)
 	// BACK TO OUTPUT PROCESSING
 	// Check if external DMA, if so, we need to be sure the read is completed
 	if (m_externalMemory && m_memory->getSlot()->isUseDma()) {
-	    // Using DMA
+	    // Using DMA so we have to busy-wait here until DMA is done
 		while (m_memory->getSlot()->isReadBusy()) {}
 	}
 
-#ifdef INTERPOLATED_DELAY
+	double bufferIndexFloat;
-	// TODO: partial delay testing
+	int delayIndex;
-	// extendedBuffer is oversized
+	for (int i=0, j=AUDIO_BLOCK_SAMPLES-1; i<AUDIO_BLOCK_SAMPLES; i++,j--) {
-	//memcpy(blockToOutput->data, &extendedBuffer[1], sizeof(int16_t)*AUDIO_BLOCK_SAMPLES);
+	    // each output sample will be an interpolated value between two samples
-	m_memory->interpolateDelay(extendedBuffer, blockToOutput->data, 0.1f, AUDIO_BLOCK_SAMPLES);
+	    // the precise delay value will be based on the LFO vector values.
-#endif
+	    // For each output sample, calculate the floating point delay offset from the reference delay.
+	    // This will be an offset from location AUDIO_BLOCK_SAMPLES/2 (e.g. 64) in the buffer.
+	    float offsetDelayFromRef = m_averageDelaySamples + (lfoValues[i] * m_delayRange) - referenceDelay;
+	    float bufferPosition = DELAY_REFERENCE_F + offsetDelayFromRef;
+
+	    // Get the interpolation coefficients from the fractional part of the buffer position
+	    float fraction1 = modf(bufferPosition, &bufferIndexFloat);
+	    float fraction2 = 1.0f -  fraction1;
+	    //fraction1 = 0.5f;
+	    //fraction2 = 0.5f;
+	    delayIndex = static_cast<unsigned>(bufferIndexFloat);
+	    if ( (delayIndex < 0) || (delayIndex > 256) ) {
+	        Serial.println(String("lfoValues[") + i + String("]:") + lfoValues[i] +
+	                       String(" referenceDelay:") + referenceDelay +
+	                       String(" bufferPosition:") + bufferPosition +
+	                       String(" delayIndex:") + delayIndex) ;
+	    }
+
+	    //delayIndex = 64+i;
+	    blockToOutput->data[j] = static_cast<int16_t>(
+	                               (static_cast<float>(extendedBuffer[j+delayIndex])   * fraction1) +
+	                               (static_cast<float>(extendedBuffer[j+delayIndex+1]) * fraction2) );
+	    //blockToOutput->data[i] = extendedBuffer[64+i];
+	}
 
 	// perform the wet/dry mix mix
 	m_postProcessing(blockToOutput, inputAudioBlock, blockToOutput);
@@ -174,6 +236,7 @@ void AudioEffectAnalogChorus::update(void)
 void AudioEffectAnalogChorus::m_preProcessing(audio_block_t *out, audio_block_t *dry, audio_block_t *wet)
 {
     memcpy(out->data, dry->data, sizeof(int16_t) * AUDIO_BLOCK_SAMPLES);
+    // TODO: Clean this up with proper preprocessing
 //	if ( out && dry && wet) {
 //		alphaBlend(out, dry, wet, m_feedback);
 //		m_iir->process(out->data, out->data, AUDIO_BLOCK_SAMPLES);
@@ -200,45 +263,25 @@ void AudioEffectAnalogChorus::m_postProcessing(audio_block_t *out, audio_block_t
 
 void AudioEffectAnalogChorus::setDelayConfig(float averageDelayMs, float delayRangeMs)
 {
-    size_t delaySamples = calcAudioSamples(averageDelayMs + delayRangeMs);
+    setDelayConfig(calcAudioSamples(averageDelayMs), calcAudioSamples(delayRangeMs));
-
-    if (delaySamples > m_memory->getMaxDelaySamples()) {
-        // this exceeds max delay value, limit it.
-        delaySamples = m_memory->getMaxDelaySamples();
-    }
-
-    if (!m_memory) { Serial.println("delay(): m_memory is not valid"); }
-
-    if (!m_externalMemory) {
-        // internal memory
-        // Do nothing
-    } else {
-        // external memory
-        ExtMemSlot *slot = m_memory->getSlot();
-
-        if (!slot) { Serial.println("ERROR: slot ptr is not valid"); }
-        if (!slot->isEnabled()) {
-            slot->enable();
-            Serial.println("WEIRD: slot was not enabled");
-        }
-    }
 }
 
 void AudioEffectAnalogChorus::setDelayConfig(size_t averageDelayNumSamples, size_t delayRangeNumSamples)
 {
     size_t delaySamples = averageDelayNumSamples + delayRangeNumSamples;
+    m_averageDelaySamples = averageDelayNumSamples;
+    m_delayRange = delayRangeNumSamples;
 
     if (delaySamples > m_memory->getMaxDelaySamples()) {
         // this exceeds max delay value, limit it.
         delaySamples = m_memory->getMaxDelaySamples();
+        m_averageDelaySamples = delaySamples/2;
+        m_delayRange = delaySamples/2;
     }
 
     if (!m_memory) { Serial.println("delay(): m_memory is not valid"); }
 
-    if (!m_externalMemory) {
+    if (m_externalMemory) {
-        // internal memory
-        // Do nothing
-    } else {
         // external memory
         ExtMemSlot *slot = m_memory->getSlot();