Hermann_Kitio
7862f7c9f3
feat(audio): pcm-capture-processor.js AudioWorklet (16kHz resample, Int16 LE) feat(hooks): useAudioCapture (getUserMedia + worklet + onChunk base64) feat(hooks): useAudioPlayback (24kHz sequential scheduling, gap-free) feat(hooks): useAudioRecording (chronological buffer, resample 16→24k, WAV export) feat(lib): audio-utils (base64, int16/float32, resample, WAV header) test: 12 audio-utils + 7 useAudioRecording = 238/238 green (+19)
80 lines
2.7 KiB
JavaScript
80 lines
2.7 KiB
JavaScript
/**
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* pcm-capture-processor.js — AudioWorklet processor pour T2 Live (Sprint 6b).
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*
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* Capture du micro à `sampleRate` natif du navigateur (typiquement 48 kHz),
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* rééchantillonnage vers 16 kHz si nécessaire, conversion Float32 → Int16
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* little-endian, envoi par chunks de ~4096 samples (≈ 256 ms à 16 kHz).
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*
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* Format de sortie attendu par Gemini Live API :
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* PCM brut, 16 kHz, 16 bits, little-endian, mono.
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*
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* Le rééchantillonnage utilise une interpolation linéaire — équivalent
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* à `resample16kTo24k` côté audio-utils.ts mais en sens inverse.
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*
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* Vanille JS (pas TS) : les AudioWorklet processors s'exécutent dans un
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* scope global isolé qui ne peut pas importer depuis le bundle TS.
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*/
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const TARGET_SAMPLE_RATE = 16000
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const CHUNK_SIZE_16K = 4096 // ≈ 256 ms à 16 kHz
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class PcmCaptureProcessor extends AudioWorkletProcessor {
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constructor() {
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super()
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this.buffer16k = new Float32Array(0)
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}
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/**
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* Rééchantillonne un Float32 du sample rate source vers 16 kHz par
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* interpolation linéaire. Si srcRate === 16000, no-op.
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*/
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resampleTo16k(input, srcRate) {
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if (srcRate === TARGET_SAMPLE_RATE) return input
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const ratio = TARGET_SAMPLE_RATE / srcRate
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const outLength = Math.floor(input.length * ratio)
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const out = new Float32Array(outLength)
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for (let i = 0; i < outLength; i++) {
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const srcIndex = i / ratio
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const srcFloor = Math.floor(srcIndex)
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const srcCeil = Math.min(srcFloor + 1, input.length - 1)
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const frac = srcIndex - srcFloor
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out[i] = input[srcFloor] * (1 - frac) + input[srcCeil] * frac
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}
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return out
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}
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process(inputs) {
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const input = inputs[0]
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if (!input || !input[0]) return true
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const channelData = input[0] // mono
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// Rééchantillonner d'abord vers 16 kHz puis accumuler.
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// `sampleRate` est une variable globale du scope AudioWorklet (Web Audio spec).
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const resampled = this.resampleTo16k(channelData, sampleRate)
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const newBuffer = new Float32Array(this.buffer16k.length + resampled.length)
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newBuffer.set(this.buffer16k)
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newBuffer.set(resampled, this.buffer16k.length)
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this.buffer16k = newBuffer
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while (this.buffer16k.length >= CHUNK_SIZE_16K) {
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const chunk = this.buffer16k.slice(0, CHUNK_SIZE_16K)
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this.buffer16k = this.buffer16k.slice(CHUNK_SIZE_16K)
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// Float32 [-1, 1] → Int16 PCM little-endian
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const pcm = new ArrayBuffer(chunk.length * 2)
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const view = new DataView(pcm)
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for (let i = 0; i < chunk.length; i++) {
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const s = Math.max(-1, Math.min(1, chunk[i]))
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view.setInt16(i * 2, s < 0 ? s * 0x8000 : s * 0x7fff, true)
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}
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this.port.postMessage(pcm, [pcm])
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}
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return true
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}
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}
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registerProcessor('pcm-capture-processor', PcmCaptureProcessor)
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