Complete rebuild of voice input on a server-authoritative streaming architecture, replacing the legacy Web Speech / whole-blob / WASM engines and the dead voice-agent layer (~4k lines removed). Speech-to-text (dictation): - Client streams 16 kHz mono PCM16 chunks over /api/dictation/ws with seq/ack ordering; buffered audio is retained and replayed on reconnect - Server transcribes and streams live partial transcripts back; segments auto-commit every ~15s with silence suppression and adaptive finalization timeouts - Local provider (default, zero config): sherpa-onnx models in a forked worker process — auto-download with progress, staged extraction with verification, corrupt-model auto-recovery, idle shutdown after 5 min - Model catalog with settings picker (accuracy/speed ratings, sizes, download/delete): Parakeet TDT v2 (English) and v3 (25 European languages, auto-detected), Whisper base and tiny (multilingual, light) - OpenAI-compatible provider for any Whisper endpoint - Composer overlay with live transcript, volume meter, timer, and cancel / insert / insert-and-send actions; failed transcriptions keep their audio for retry or accepting the partial text as-is - Configurable keyboard shortcut (default mod+alt+v) toggles dictation; Enter confirms and Escape cancels while recording - Overlay is pixel-aligned with the composer (measured footer height, matching paddings/typography/gaps) — no layout shift when toggling Text-to-speech: - Local Kokoro provider (English, 11 voices) synthesized in the same worker via /api/dictation/tts/speak, managed by the shared model pipeline; sentence-pipelined playback keeps time-to-first-audio at ~1 sentence regardless of message length, and stop cancels in-flight synthesis - Sanitizer keeps inline-code content (strips backticks only), reads interword slashes aloud, and removes only absolute file paths Settings: - Voice page unified: a single read-aloud toggle owns all playback options (the confusing "Enable Voice Mode" is gone); a new "Enable voice input" toggle (default on, persisted to settings.json) hides the composer mic entirely when disabled Mobile and transport: - iOS/Android microphone permissions added (dictation was previously impossible on mobile) - Fixed Android WebSocket upgrades: the Capacitor WebView origin (https://localhost) was missing from the packaged-client allowlist, 403-ing every WS connection — root cause of the old mobile SSE lock, which is now removed for all transports Security and conventions: - All HTTP routes sit behind the global /api auth gate; the WS upgrade explicitly validates the UI session and origin, with oc_url_token narrowly allowlisted and covered by tests; the dictation socket mints a fresh URL token before connecting - Routes register before the generic OpenCode proxy; the client goes through runtimeFetch/getRuntimeUrlResolver, and runtime switches reset the dictation socket - VS Code deliberately reports dictation as unavailable (no server process in that runtime) CI: workflow Node bumped 20 -> 22 to match the repo engines and fix better-sqlite3 installs broken by node-gyp@latest on Node 20. New dependency: sherpa-onnx-node (prebuilt N-API; macOS/Linux x64+arm64, Windows x64 — Windows-on-ARM falls back to the OpenAI-compatible provider)
267 lines
8.8 KiB
TypeScript
267 lines
8.8 KiB
TypeScript
/**
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* useLocalTTS Hook
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*
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* React hook for local (Kokoro via sherpa-onnx) text-to-speech playback.
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* Synthesis runs on the OpenChamber server in the dictation worker.
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*
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* Long texts are pipelined by sentence chunks: the first chunk starts playing
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* as soon as it is synthesized while the next chunk synthesizes in the
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* background, so time-to-first-audio stays ~1 chunk regardless of message
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* length.
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*/
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import { useCallback, useEffect, useRef, useState } from 'react';
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import { runtimeFetch } from '@/lib/runtime-fetch';
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export interface LocalTTSSpeakOptions {
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/** Kokoro speaker id (0-10) */
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speakerId?: number;
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/** Playback speed multiplier (1.0 = normal) */
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speed?: number;
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onStart?: () => void;
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onEnd?: () => void;
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onError?: (error: string) => void;
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}
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export interface UseLocalTTSReturn {
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isPlaying: boolean;
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error: string | null;
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speak: (text: string, options?: LocalTTSSpeakOptions) => Promise<void>;
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stop: () => void;
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/** Unlock audio for mobile Safari - call this on user gesture */
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unlockAudio: () => Promise<void>;
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}
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/** Target chunk size: big enough to amortize requests, small enough for low latency. */
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const MIN_CHUNK_CHARS = 60;
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const MAX_CHUNK_CHARS = 400;
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/**
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* Split text into sentence-aligned chunks for pipelined synthesis.
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* Sentences are merged until MIN_CHUNK_CHARS and hard-split at
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* MAX_CHUNK_CHARS so a single run-on sentence cannot stall the pipeline.
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*/
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export function splitTextForSynthesis(text: string): string[] {
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const normalized = text.replace(/\s+/g, ' ').trim();
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if (!normalized) {
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return [];
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}
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const sentences = normalized.match(/[^.!?…]+[.!?…]+["')\]]*\s*|[^.!?…]+$/g) ?? [normalized];
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const chunks: string[] = [];
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let current = '';
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for (const sentence of sentences) {
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for (let offset = 0; offset < sentence.length; offset += MAX_CHUNK_CHARS) {
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const piece = sentence.slice(offset, offset + MAX_CHUNK_CHARS);
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if (current && current.length + piece.length > MAX_CHUNK_CHARS) {
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chunks.push(current.trim());
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current = '';
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}
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current += piece;
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if (current.length >= MIN_CHUNK_CHARS && /[.!?…]["')\]]*\s*$/.test(current)) {
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chunks.push(current.trim());
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current = '';
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}
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}
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}
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if (current.trim()) {
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chunks.push(current.trim());
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}
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// Latency trim for the FIRST chunk only: if the opening sentence is long,
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// split it at a clause boundary (comma/semicolon/colon) so the first
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// audio starts sooner. Only the first chunk gets this treatment — comma
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// splits inside every sentence would chop the prosody, and after playback
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// starts the pipeline hides synthesis time anyway.
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if (chunks.length > 0 && chunks[0].length > 120) {
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const first = chunks[0];
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const clauseBreak = /[,;:]\s/g;
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let bestBreak = -1;
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let match: RegExpExecArray | null;
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while ((match = clauseBreak.exec(first)) !== null) {
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const end = match.index + 1;
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if (end < 40) {
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continue;
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}
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if (end > 160) {
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break;
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}
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bestBreak = end;
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break;
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}
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if (bestBreak > 0) {
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const head = first.slice(0, bestBreak).trim();
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const tail = first.slice(bestBreak).trim();
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chunks.splice(0, 1, head, tail);
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}
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}
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return chunks;
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}
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let sharedAudioContext: AudioContext | null = null;
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function getAudioContext(): AudioContext {
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if (!sharedAudioContext) {
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sharedAudioContext = new (window.AudioContext || (window as unknown as { webkitAudioContext: typeof AudioContext }).webkitAudioContext)();
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}
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return sharedAudioContext;
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}
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interface PlaybackSession {
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cancelled: boolean;
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abort: AbortController;
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}
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export function useLocalTTS(): UseLocalTTSReturn {
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const [isPlaying, setIsPlaying] = useState(false);
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const [error, setError] = useState<string | null>(null);
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const audioSourceRef = useRef<AudioBufferSourceNode | null>(null);
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const sessionRef = useRef<PlaybackSession | null>(null);
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const unlockAudio = useCallback(async (): Promise<void> => {
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try {
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const ctx = getAudioContext();
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if (ctx.state === 'suspended') {
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await ctx.resume();
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}
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const buffer = ctx.createBuffer(1, 1, 22050);
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const source = ctx.createBufferSource();
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source.buffer = buffer;
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source.connect(ctx.destination);
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source.start(0);
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} catch {
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// Unlocking is best-effort.
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}
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}, []);
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const stop = useCallback(() => {
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const session = sessionRef.current;
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if (session) {
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session.cancelled = true;
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session.abort.abort();
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sessionRef.current = null;
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}
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if (audioSourceRef.current) {
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try {
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audioSourceRef.current.onended = null;
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audioSourceRef.current.stop();
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} catch {
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// Already stopped
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}
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audioSourceRef.current = null;
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}
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setIsPlaying(false);
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}, []);
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const speak = useCallback(async (text: string, options?: LocalTTSSpeakOptions): Promise<void> => {
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stop();
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const chunks = splitTextForSynthesis(text);
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if (chunks.length === 0) {
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setError('No text to speak');
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options?.onError?.('No text to speak');
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return;
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}
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setError(null);
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const session: PlaybackSession = { cancelled: false, abort: new AbortController() };
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sessionRef.current = session;
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const fetchChunk = async (chunk: string): Promise<ArrayBuffer> => {
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const response = await runtimeFetch('/api/dictation/tts/speak', {
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method: 'POST',
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headers: { 'Content-Type': 'application/json' },
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body: JSON.stringify({
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text: chunk,
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...(typeof options?.speakerId === 'number' ? { speakerId: options.speakerId } : {}),
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...(typeof options?.speed === 'number' ? { speed: options.speed } : {}),
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}),
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signal: session.abort.signal,
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});
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if (!response.ok) {
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const errorData = await response.json().catch(() => ({ error: 'Unknown error' }));
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throw new Error(errorData.error || `HTTP ${response.status}`);
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}
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return (await response.blob()).arrayBuffer();
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};
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const playBuffer = async (arrayBuffer: ArrayBuffer): Promise<void> => {
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const ctx = getAudioContext();
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if (ctx.state === 'suspended') {
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await ctx.resume();
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}
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const audioBuffer = await ctx.decodeAudioData(arrayBuffer);
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if (session.cancelled) {
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return;
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}
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await new Promise<void>((resolve) => {
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const source = ctx.createBufferSource();
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source.buffer = audioBuffer;
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source.connect(ctx.destination);
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audioSourceRef.current = source;
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source.onended = () => {
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if (audioSourceRef.current === source) {
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audioSourceRef.current = null;
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}
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resolve();
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};
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source.start(0);
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});
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};
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try {
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setIsPlaying(true);
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options?.onStart?.();
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// Pipelined: synthesize chunk N+1 while chunk N is playing.
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let nextFetch: Promise<ArrayBuffer> = fetchChunk(chunks[0]);
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for (let i = 0; i < chunks.length; i += 1) {
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const buffer = await nextFetch;
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if (session.cancelled) {
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return;
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}
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if (i + 1 < chunks.length) {
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nextFetch = fetchChunk(chunks[i + 1]);
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}
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await playBuffer(buffer);
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if (session.cancelled) {
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return;
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}
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}
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setIsPlaying(false);
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options?.onEnd?.();
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} catch (err) {
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if ((err as Error).name === 'AbortError' || session.cancelled) {
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return;
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}
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const errorMsg = err instanceof Error ? err.message : 'Failed to speak';
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setError(errorMsg);
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options?.onError?.(errorMsg);
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setIsPlaying(false);
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} finally {
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if (sessionRef.current === session) {
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sessionRef.current = null;
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}
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}
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}, [stop]);
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useEffect(() => {
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return () => {
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stop();
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};
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}, [stop]);
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return {
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isPlaying,
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error,
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speak,
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stop,
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unlockAudio,
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};
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}
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