Buffering of audio

AudioThread.cpp

@@ -26,6 +26,12 @@ double phase = 0.0;
 unsigned long lastNoteChangeTime = 0;
 // ---
 
+// Ring Buffer
+int16_t audioBuffer[AUDIO_BUFFER_SIZE];
+int audioHead = 0;
+int audioTail = 0;
+bool audioBuffering = true;
+
 void setupAudio() {
   // Configure I2S pins
   i2s.setBCLK(I2S_BCLK_PIN);
@@ -43,7 +49,9 @@ void setupAudio() {
   
   // Initialize the portable synth engine
   globalSynth = new SynthEngine(SAMPLE_RATE);
-  globalSynth->loadPreset(2);
+  if (globalSynth) {
+    globalSynth->loadPreset(2);
+  }
 }
 
 void loopAudio() {
@@ -71,38 +79,43 @@ void loopAudio() {
     }
   }
 
-  const int BATCH_SIZE = 16;
-
-  // Ensure we don't generate samples faster than the I2S can consume them.
-  // We write 2 samples (Left + Right) for every 1 synth sample.
-  if (i2s.availableForWrite() < BATCH_SIZE * 2) {
-      return;
-  }
-
-  // Process a small batch of samples
-  static int16_t samples[BATCH_SIZE];
-
-  // Generate sound samples
-  if (globalSynth) {
-    // using synth engine
-    globalSynth->process(samples, BATCH_SIZE);
+  // Produce samples in a cyclic buffer
+  int nextHead = (audioHead + 1) % AUDIO_BUFFER_SIZE;
+  if (nextHead != audioTail) {
+      if (audioHead == audioTail) {
+          audioBuffering = true;
+      }
+      int16_t sample = 0;
+      if (globalSynth) {
+          globalSynth->process(&sample, 1);
+      } else {
+          if (currentFrequency > 0) {
+              phase += 2.0 * M_PI * currentFrequency / SAMPLE_RATE;
+              if (phase >= 2.0 * M_PI) phase -= 2.0 * M_PI;
+              sample = phase * 0.1f * AMPLITUDE;
+          } else {
+              sample = 0;
+          }
+      }
+      audioBuffer[audioHead] = sample;
+      audioHead = nextHead;
   } else {
-    // using fallback sawtooth
-    for (int i = 0; i < BATCH_SIZE; ++i) {
-        if (currentFrequency > 0) {
-            //samples[i] = (int16_t)(sin(phase) * AMPLITUDE);
-            phase += 2.0 * M_PI * currentFrequency / SAMPLE_RATE;
-            if (phase >= 2.0 * M_PI) phase -= 2.0 * M_PI;
-            samples[i] = phase * 0.1f * AMPLITUDE;
-        } else {
-            samples[i] = 0;
-        }
-    }
+    audioBuffering = false;
   }
 
-  // write out stereo samples
-  for (int i = 0; i < BATCH_SIZE; ++i) {
-      i2s.write(samples[i]);
-      i2s.write(samples[i]);
+  // Consume samples from this buffer whenever there is capacity in i2s
+  while (!audioBuffering && audioHead != audioTail) {
+      if (i2s.availableForWrite() < 2) {
+          break;
+      }
+      int16_t s = audioBuffer[audioTail];
+      i2s.write(s);
+      i2s.write(s);
+      audioTail = (audioTail + 1) % AUDIO_BUFFER_SIZE;
   }
+
+  // Update usage stats
+  int usage = audioHead - audioTail;
+  if (usage < 0) usage += AUDIO_BUFFER_SIZE;
+  audioBufferUsage = usage;
 }