Added intepolated delay capability to AudioDelay class

examples/Tests/TGA_PRO_MEM2_EXP/TGA_PRO_MEM2_EXP.ino

@@ -85,6 +85,7 @@ unsigned loopCounter = 0;
 
 void setup() {
   Serial.begin(57600);
+  delay(500);
 
   // Disable the audio codec first
   codec.disable();

src/LibBasicFunctions.h

@@ -168,6 +168,9 @@ public:
     /// @param numSamples default value is AUDIO_BLOCK_SAMPLES, so typically you don't have to specify this parameter.
     /// @returns true on success, false on error.
     bool getSamples(audio_block_t *dest, size_t offsetSamples, size_t numSamples = AUDIO_BLOCK_SAMPLES);
+    bool getSamples(int16_t *dest, size_t offsetSamples, size_t numSamples);
+
+    bool interpolateDelay(int16_t *extendedSourceBuffer, int16_t *destBuffer, float fraction, 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.
@@ -193,6 +196,7 @@ private:
     RingBuffer<audio_block_t *> *m_ringBuffer = nullptr; ///< When using INTERNAL memory, a RingBuffer will be created.
     ExtMemSlot *m_slot = nullptr;                        ///< When using EXTERNAL memory, an ExtMemSlot must be provided.
     size_t m_maxDelaySamples = 0;                        ///< stores the number of audio samples in the AudioDelay.
+    bool m_getSamples(int16_t *dest, size_t offsetSamples, size_t numSamples); ///< operates directly on int16_y buffers
 };
 
 /**************************************************************************//**

src/common/AudioDelay.cpp

@@ -17,6 +17,7 @@
  *  You should have received a copy of the GNU General Public License
  *  along with this program.  If not, see <http://www.gnu.org/licenses/>.
 */
+#include <cmath>
 
 #include "Audio.h"
 #include "LibBasicFunctions.h"
@@ -105,6 +106,16 @@ size_t AudioDelay::getMaxDelaySamples()
 }
 
 bool AudioDelay::getSamples(audio_block_t *dest, size_t offsetSamples, size_t numSamples)
+{
+	return m_getSamples(dest->data, offsetSamples, numSamples);
+}
+
+bool AudioDelay::getSamples(int16_t *dest, size_t offsetSamples, size_t numSamples)
+{
+	return m_getSamples(dest, offsetSamples, numSamples);
+}
+
+bool AudioDelay::m_getSamples(int16_t *dest, size_t offsetSamples, size_t numSamples)
 {
 	if (!dest) {
 		Serial.println("getSamples(): dest is invalid");
@@ -120,53 +131,56 @@ bool AudioDelay::getSamples(audio_block_t *dest, size_t offsetSamples, size_t nu
 		audio_block_t *currentQueue1 = m_ringBuffer->at(m_ringBuffer->get_index_from_back(index+1));
 
 		// check if either queue is invalid, if so just zero the destination buffer
-		if ( (!currentQueue0) || (!currentQueue0) ) {
+		if ( (!currentQueue0) || (!currentQueue1) ) {
 			// a valid entry is not in all queue positions while it is filling, use zeros
-			memset(static_cast<void*>(dest->data), 0, numSamples * sizeof(int16_t));
+			memset(static_cast<void*>(dest), 0, numSamples * sizeof(int16_t));
 			return true;
 		}
 
-		if (position.offset == 0) {
+		if ( (position.offset == 0) && numSamples <= AUDIO_BLOCK_SAMPLES ) {
 			// single transfer
-			memcpy(static_cast<void*>(dest->data), static_cast<void*>(currentQueue0->data), numSamples * sizeof(int16_t));
+			memcpy(static_cast<void*>(dest), static_cast<void*>(currentQueue0->data), numSamples * sizeof(int16_t));
 			return true;
 		}
 
 		// Otherwise we need to break the transfer into two memcpy because it will go across two source queues.
 		// Audio is stored in reverse order. That means the first sample (in time) goes in the last location in the audio block.
-		int16_t *destStart = dest->data;
+		int16_t *destStart = dest;
 		int16_t *srcStart;
 
 		// Break the transfer into two. Copy the 'older' data first then the 'newer' data with respect to current time.
-		//currentQueue = m_ringBuffer->at(m_ringBuffer->get_index_from_back(index+1)); // The latest buffer is at the back. We need index+1 counting from the back.
+		// TODO: should AUDIO_BLOCK_SAMPLES on the next line be numSamples?
 		srcStart  = (currentQueue1->data + AUDIO_BLOCK_SAMPLES - position.offset);
 		size_t numData = position.offset;
 		memcpy(static_cast<void*>(destStart), static_cast<void*>(srcStart), numData * sizeof(int16_t));
 
-		//currentQueue = m_ringBuffer->at(m_ringBuffer->get_index_from_back(index)); // now grab the queue where the 'first' data sample was
 		destStart += numData; // we already wrote numData so advance by this much.
 		srcStart  = (currentQueue0->data);
-		numData = AUDIO_BLOCK_SAMPLES - numData;
+		numData = numSamples - numData;
 		memcpy(static_cast<void*>(destStart), static_cast<void*>(srcStart), numData * sizeof(int16_t));
 
 		return true;
 
 	} else {
 		// EXTERNAL Memory
-		if (numSamples*sizeof(int16_t) <= m_slot->size() ) {
+		if (numSamples*sizeof(int16_t) <= m_slot->size() ) { // check for overflow
-			int currentPositionBytes = (int)m_slot->getWritePosition() - (int)(AUDIO_BLOCK_SAMPLES*sizeof(int16_t));
+			// current position is considered the write position subtracted by the number of samples we're going
+			// to read since this is the smallest delay we can get without reading past the write position into
+			// the "future".
+			int currentPositionBytes = (int)m_slot->getWritePosition() - (int)(numSamples*sizeof(int16_t));
 			size_t offsetBytes = offsetSamples * sizeof(int16_t);
 
 			if ((int)offsetBytes <= currentPositionBytes) {
+				// when we back up to read, we won't wrap over the beginning of the slot
 				m_slot->setReadPosition(currentPositionBytes - offsetBytes);
 			} else {
-				// It's going to wrap around to the end of the slot
+				// It's going to wrap around to the from the beginning to the end of the slot.
 				int readPosition = (int)m_slot->size() + currentPositionBytes - offsetBytes;
 				m_slot->setReadPosition((size_t)readPosition);
 			}
 
-			// This causes pops
+			// Read the number of samples
-			m_slot->readAdvance16(dest->data, AUDIO_BLOCK_SAMPLES);
+			m_slot->readAdvance16(dest, numSamples);
 
 			return true;
 		} else {
@@ -179,5 +193,18 @@ bool AudioDelay::getSamples(audio_block_t *dest, size_t offsetSamples, size_t nu
 
 }
 
+bool AudioDelay::interpolateDelay(int16_t *extendedSourceBuffer, int16_t *destBuffer, float fraction, size_t numSamples)
+{
+	int16_t frac1 = static_cast<int16_t>(32767.0f * fraction);
+	int16_t frac2 = 32767 - frac1;
+
+	// TODO optimize this later
+	for (int i=0; i<numSamples; i++) {
+		destBuffer[i] =  ((frac1*extendedSourceBuffer[i]) >> 16) +  ((frac2*extendedSourceBuffer[i+1]) >> 16);
+	}
+	return true;
+}
+
+
 }
 

src/effects/AudioEffectAnalogDelay.cpp

@@ -10,6 +10,8 @@
 
 using namespace BALibrary;
 
+#define INTERPOLATED_DELAY
+
 namespace BAEffects {
 
 constexpr int MIDI_CHANNEL = 0;
@@ -124,7 +126,13 @@ void AudioEffectAnalogDelay::update(void)
 
     // get the data. If using external memory with DMA, this won't be filled until
     // later.
+#ifdef INTERPOLATED_DELAY
+    int16_t extendedBuffer[AUDIO_BLOCK_SAMPLES+1]; // need one more sample for intepolating between 128th and 129th (last sample)
+    m_memory->getSamples(extendedBuffer, m_delaySamples, AUDIO_BLOCK_SAMPLES+1);
+#else
     m_memory->getSamples(blockToOutput, m_delaySamples);
+#endif
+
 
     // If using DMA, we need something else to do while that read executes, so
     // move on to input preprocessing
@@ -146,6 +154,13 @@ void AudioEffectAnalogDelay::update(void)
 		while (m_memory->getSlot()->isReadBusy()) {}
 	}
 
+#ifdef INTERPOLATED_DELAY
+	// TODO: partial delay testing
+	// extendedBuffer is oversized
+	//memcpy(blockToOutput->data, &extendedBuffer[1], sizeof(int16_t)*AUDIO_BLOCK_SAMPLES);
+	m_memory->interpolateDelay(extendedBuffer, blockToOutput->data, 0.1f, AUDIO_BLOCK_SAMPLES);
+#endif
+
 	// perform the wet/dry mix mix
 	m_postProcessing(blockToOutput, inputAudioBlock, blockToOutput);
 	transmit(blockToOutput);