fix 50ms and update examples

examples/publish-wav/index.ts

@@ -44,12 +44,11 @@ const track = LocalAudioTrack.createAudioTrack('audio', source);
 const options = new TrackPublishOptions();
 const buffer = new Int16Array(sample.buffer);
 options.source = TrackSource.SOURCE_MICROPHONE;
-await room.localParticipant.publishTrack(track, options);
-await new Promise((resolve) => setTimeout(resolve, 1000)); // wait a bit so the start doesn't cut off
+await room.localParticipant.publishTrack(track, options).then((pub) => pub.waitForSubscription());
 
 let written = 44; // start of WAVE data stream
 const FRAME_DURATION = 1; // write 1s of audio at a time
-const numSamples = sampleRate / FRAME_DURATION;
+const numSamples = sampleRate * FRAME_DURATION;
 while (written < dataSize) {
   const available = dataSize - written;
   const frameSize = Math.min(numSamples, available);
@@ -61,6 +60,7 @@ while (written < dataSize) {
     Math.trunc(frameSize / channels),
   );
   await source.captureFrame(frame);
+  await source.waitForPlayout();
 
   written += frameSize;
 }

packages/livekit-rtc/src/audio_source.ts

@@ -43,7 +43,10 @@ export class AudioSource {
     this.queueSize = queueSize;
 
     this.lastCapture = 0;
-    this.currentQueueSize = 0;
+
+    // remove 50ms to account for processing time
+    // (e.g. using wait_for_playout for very small chunks)
+    this.currentQueueSize = -50;
 
     const req = new NewAudioSourceRequest({
       type: AudioSourceType.AUDIO_SOURCE_NATIVE,
@@ -97,8 +100,6 @@ export class AudioSource {
     const elapsed = this.lastCapture === 0 ? 0 : now - this.lastCapture;
     this.currentQueueSize += (frame.samplesPerChannel / frame.sampleRate - elapsed) * 1000;
 
-    // remove 50ms to account for processing time (e.g. using wait_for_playout for very small chunks)
-    this.currentQueueSize -= 50;
     this.lastCapture = now;
 
     if (this.timeout) {