Audio fixes

- Use smaller buffer to avoid too much audio after wait=True
- Don't reset nextStartTime to avoid overlapping audio
- Add indirection to callback so we can clear it properly to avoid
  calling it when audio finishes after MicroPython has terminated

src/board/audio/index.ts

@@ -30,9 +30,7 @@ export class BoardAudio {
   currentSoundExpressionCallback: undefined | (() => void);
   private stopActiveRecording: (() => void) | undefined;
 
-  constructor(
-    private microphoneEl: SVGElement
-  ) {}
+  constructor(private microphoneEl: SVGElement) {}
 
   initializeCallbacks({
     defaultAudioCallback,
@@ -142,9 +140,9 @@ export class BoardAudio {
   setSensitivity(sensitivity: number) {
     this.sensitivityNode!.gain.setValueAtTime(
       // check if this is correct
-      sensitivity, 
+      sensitivity,
       this.context!.currentTime
-    )
+    );
   }
 
   setVolume(volume: number) {
@@ -206,7 +204,7 @@ export class BoardAudio {
       this.stopRecording();
       return;
     }
-    this.microphoneEl.style.display = "unset"
+    this.microphoneEl.style.display = "unset";
 
     const source = this.context!.createMediaStreamSource(micStream);
     source.connect(this.sensitivityNode!);
@@ -241,8 +239,8 @@ export class BoardAudio {
       recorder.disconnect();
       this.sensitivityNode!.disconnect();
       source.disconnect();
-      micStream.getTracks().forEach(track => track.stop())
-      this.microphoneEl.style.display = "none"
+      micStream.getTracks().forEach((track) => track.stop());
+      this.microphoneEl.style.display = "none";
       this.stopActiveRecording = undefined;
     };
   }
@@ -274,8 +272,9 @@ class BufferedAudio {
   ) {}
 
   init(sampleRate: number) {
+    // This is called for each new audio source so don't reset nextStartTime
+    // or we start to overlap audio
     this.sampleRate = sampleRate;
-    this.nextStartTime = -1;
   }
 
   createBuffer(length: number) {
@@ -291,20 +290,24 @@ class BufferedAudio {
     // Use createBufferSource instead of new AudioBufferSourceNode to support Safari 14.0.
     const source = this.context.createBufferSource();
     source.buffer = buffer;
-    source.onended = this.callback;
+    source.onended = this.callCallback;
     source.connect(this.destination);
     const currentTime = this.context.currentTime;
     const first = this.nextStartTime < currentTime;
     const startTime = first ? currentTime : this.nextStartTime;
     this.nextStartTime = startTime + buffer.length / buffer.sampleRate;
-    // For audio frames, we're frequently out of data. Speech is smooth.
     if (first) {
       // We're just getting started so buffer another frame.
       this.callback();
     }
     source.start(startTime);
   }
 
+  private callCallback = () => {
+    // Indirect so we can clear callback later
+    this.callback();
+  };
+
   dispose() {
     // Prevent calls into WASM when the buffer nodes finish.
     this.callback = () => {};

src/jshal.js

@@ -278,10 +278,8 @@ mergeInto(LibraryManager.library, {
     Module.board.microphone.microphoneOn();
   },
 
-  mp_js_hal_microphone_set_sensitivity: function (
-    /** @type {number} */ value
-  ) {
-    Module.board.audio.setSensitivity(value)
+  mp_js_hal_microphone_set_sensitivity: function (/** @type {number} */ value) {
+    Module.board.audio.setSensitivity(value);
   },
   mp_js_hal_microphone_set_threshold: function (
     /** @type {number} */ kind,

src/microbithal_js.c

@@ -455,7 +455,16 @@ void microbit_hal_audio_raw_set_rate(uint32_t sample_rate) {
 }
 
 void microbit_hal_audio_raw_write_data(const uint8_t *buf, size_t num_samples) {
-    mp_js_hal_audio_write_data(buf, num_samples);
+    bool silence = true;
+    for (const uint8_t *sample = buf; sample < buf + num_samples; ++sample) {
+        if (*sample != 128) {
+            silence = false;
+            break;
+        }
+    }
+    if (!silence) {
+        mp_js_hal_audio_write_data(buf, num_samples);
+    }
 }
 
 void microbit_hal_audio_speech_init(uint32_t sample_rate) {

src/mpconfigport.h

@@ -129,6 +129,6 @@ extern uint32_t rng_generate_random_word(void);
     ((mp_raise_NotImplementedError(MP_ERROR_TEXT("simulator limitation: asm_thumb code"))), p)
 
 // The latency of fetching 32 byte audio frames is too much so increase the size
-#define AUDIO_OUTPUT_BUFFER_SIZE (128)
+#define AUDIO_OUTPUT_BUFFER_SIZE (64)
 
 #endif