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Improve audio capture chunking for more stable STT.

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Increase captured WebRTC audio chunk duration to reduce fragmented transcriptions and improve recognition quality in meetings.

Made-with: Cursor
This commit is contained in:
ValueOn AG 2026-02-26 21:18:07 +01:00
parent 77aa4f3d07
commit 3863063984
1 changed files with 3 additions and 2 deletions
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5
src/bot/audioCaptureProcedure.ts
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@ -95,8 +95,9 @@ export class AudioCaptureProcedure {
let samplesCollected = 0;
let skippedSilentChunks = 0;
const minRmsThreshold = 0.0015;
// Collect ~1 second of audio at native rate before emitting
const samplesPerChunk = nativeRate;
// Collect ~2 seconds of audio at native rate before emitting.
// Larger chunks improve STT stability and reduce fragment transcripts.
const samplesPerChunk = nativeRate * 2;
const targetRate = 16000;
processor.onaudioprocess = (e: AudioProcessingEvent) => {