Files
openchamber/packages/web/server/lib/dictation/openai-compatible-session.js
T
Bohdan Triapitsyn de1b85ac56 feat(voice): first-class voice input and local TTS across web, desktop, and mobile (#2018)
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)
2026-07-04 02:48:07 +03:00

99 lines
2.8 KiB
JavaScript

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