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)
99 lines
2.8 KiB
JavaScript
99 lines
2.8 KiB
JavaScript
/**
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* Pseudo-streaming transcription session for OpenAI-compatible Whisper
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* endpoints (faster-whisper, whisper.cpp, OpenAI, ...).
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*
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* The Whisper HTTP API cannot stream, so audio is buffered per segment and
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* transcribed on commit(). Live partials therefore only advance at segment
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* boundaries (the DictationStreamManager auto-commits every ~15s of speech).
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*
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* Implements the StreamingTranscriptionSession contract used by
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* DictationStreamManager.
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*/
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import { EventEmitter } from 'events';
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import { randomUUID } from 'crypto';
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import { transcribeAudio } from '../tts/stt.js';
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import { pcm16ToWav } from './audio.js';
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const OPENAI_COMPATIBLE_SAMPLE_RATE = 16000;
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export class OpenAICompatibleTranscriptionSession extends EventEmitter {
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/**
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* @param {{ baseURL: string, model: string, apiKey?: string, language?: string, prompt?: string }} config
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*/
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constructor(config) {
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super();
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this.config = config;
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this.requiredSampleRate = OPENAI_COMPATIBLE_SAMPLE_RATE;
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this.connected = false;
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this.segmentId = randomUUID();
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this.previousSegmentId = null;
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this.pcm16 = Buffer.alloc(0);
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}
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async connect() {
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if (!this.config.baseURL) {
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throw new Error('Custom STT server URL is not configured');
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}
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if (!this.config.model) {
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throw new Error('STT model is not configured');
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}
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this.connected = true;
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}
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appendPcm16(chunk) {
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if (!this.connected) {
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this.emit('error', new Error('STT session not connected'));
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return;
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}
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this.pcm16 = this.pcm16.length === 0 ? chunk : Buffer.concat([this.pcm16, chunk]);
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}
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commit() {
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if (!this.connected) {
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this.emit('error', new Error('STT session not connected'));
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return;
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}
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const committedId = this.segmentId;
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const previousSegmentId = this.previousSegmentId;
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const committedPcm16 = this.pcm16;
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this.previousSegmentId = committedId;
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this.segmentId = randomUUID();
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this.pcm16 = Buffer.alloc(0);
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this.emit('committed', { segmentId: committedId, previousSegmentId });
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void (async () => {
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try {
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const wav = pcm16ToWav(committedPcm16, OPENAI_COMPATIBLE_SAMPLE_RATE);
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const text = await transcribeAudio({
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audioBuffer: wav,
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mimeType: 'audio/wav',
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model: this.config.model,
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baseURL: this.config.baseURL,
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apiKey: this.config.apiKey,
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language: this.config.language,
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});
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this.emit('transcript', {
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segmentId: committedId,
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transcript: (text ?? '').trim(),
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isFinal: true,
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});
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} catch (err) {
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this.emit('error', err instanceof Error ? err : new Error(String(err)));
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}
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})();
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}
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clear() {
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this.pcm16 = Buffer.alloc(0);
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this.segmentId = randomUUID();
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}
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close() {
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this.connected = false;
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this.pcm16 = Buffer.alloc(0);
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}
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}
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