Ship live translations
with confidence
A production-ready full-stack Node.js + React application for seamless EN↔RU↔UK live auto-detect translation with voice synthesis.
⚙
Installation
Set up the project locally with Docker, Redis, and LibreTranslate in minutes.
▦
Architecture
Understand the STT → Translation → TTS pipeline and real-time Socket.io communication.
▶
Live Translation
Stream from YouTube or microphone with automatic EN/RU/UK language detection and voice output.
📜
Biblical Simulator
Test the full pipeline with AI-generated biblical passages in King James, Church Slavonic, or Ukrainian style.
🎤
Voice Training
Clone custom voices from microphone recordings or YouTube videos using ElevenLabs IVC.
Prerequisites
●
Node.js 20+
Runtime for backend and build tools
●
Docker + Docker Compose
For Redis and LibreTranslate services
●
yt-dlp + ffmpeg
Required for YouTube audio extraction
●
ElevenLabs API Key
For speech-to-text and text-to-speech
Clone & Configure
git clone https://github.com/Pzharyuk/live-translator-node.git && cd live-translator-node
cp .env.example .env
Edit .env and set your API key:
ELEVENLABS_API_KEY=sk-your-key-here
ADMIN_PASSWORD=your-secure-password
Start Infrastructure
# Start Redis + LibreTranslate
docker compose -f docker-compose.local.yml up -d
# Wait for LibreTranslate to download language models (~500 MB)
docker logs -f $(docker ps -qf "name=libretranslate") 2>&1 | grep -i "running"
Start Backend
cd backend
npm install
npm run dev # nodemon watches for changes
Start Frontend
cd frontend
npm install
npm run dev # Vite hot-reload on localhost:5173
✓
You're all set!
Open http://localhost:5173 for the user view and http://localhost:5173/admin for the admin panel (default password: admin123).
1
Start the services
Follow the Installation guide to get Docker services, backend, and frontend running.
2
Open the Admin Panel
Navigate to http://localhost:5173/admin and enter the admin password.
3
Select a Voice
Choose a TTS voice from the dropdown. The voice list is fetched from your ElevenLabs account.
4
Test with Text
Use the free-text area in the admin panel to type a phrase. Click translate to hear the TTS output instantly.
5
Go Live
Open the user view at http://localhost:5173. Select "Mic" as input, pick a voice, and click Start. Speak into your microphone and watch real-time translation appear with audio playback.
💡
Try the Biblical Simulator
For a hands-free demo, enable the biblical_simulator feature flag in admin, enter an Anthropic API key, select a language, and click "Generate". The system will produce biblical passages through the full STT → Translation → TTS pipeline.
System Overview
Frontend
React 19 + Vite
Socket.io Client
Web Audio API
↔
Backend
Express + Socket.io
TypeScript
Service Layer
ElevenLabs
Scribe v2 (STT)
TTS Streaming
Voice Cloning
Translation
LibreTranslate (self-hosted)
DeepL (premium API)
Claude / Anthropic (AI)
Redis
Feature Flags
Settings Store
Anthropic
Biblical Simulator
Claude Translation
DeepL
Free & Pro tiers
Auto endpoint detection
Data Flow
1 Audio Input (Mic / YouTube / Simulator)
↓
2 PCM 16-bit LE @ 16kHz via Socket.io chunks
↓
3 ElevenLabs Scribe v2 WebSocket STT
↓
4 Commit Merge Buffer 2.5s VAD aggregation
↓
5 Translation Provider LibreTranslate / DeepL / Claude
↓
6 ElevenLabs TTS Voice synthesis streaming
↓
7 Audio Playback Queued with 600ms pause
Key Architecture Decisions
Two-layer Language Detection
LibreTranslate's /detect endpoint returns 0-confidence for short Cyrillic phrases. The app uses script-based pre-detection (Unicode 0x0400–0x04FF = Cyrillic) combined with ElevenLabs Scribe's language_code output for reliable EN/RU/UK auto-detection.
VAD Commit Merging
Voice Activity Detection can fire aggressively on speaker breathing. Commits are buffered for 2.5 seconds before translation to merge fragments into meaningful phrases.
Feature Flag Merging
YAML config defaults are merged with Redis runtime overrides. Redis values take priority, falling back to YAML if Redis is unavailable.
API Key Hierarchy
Keys resolve in order: Runtime Cache → Redis → Config File → Empty. This allows hot-swapping keys without restarts.
Connection Lifecycle
- Client sends
start_session with source type (mic or youtube) and optional voiceId
- Backend opens a WebSocket to
wss://api.elevenlabs.io/v1/speech-to-text/realtime
- For YouTube: spawns
yt-dlp | ffmpeg child processes to extract PCM audio
- For Microphone: awaits
audio_chunk events from the frontend
Audio Streaming
Audio chunks are sent to Scribe as JSON messages:
{
"message_type": "input_audio_chunk",
"audio_base_64": "UklGR..." // PCM 16-bit LE, 16kHz, mono
}
Scribe Responses
| Response Type | Meaning | Action |
partial_transcript |
Live partial text (speculative) |
Emitted as non-final transcript event |
committed_transcript |
VAD fired — complete phrase |
Buffered for commit merge window |
Commit Merge Buffer
After receiving a committed_transcript, the backend waits 2.5 seconds (COMMIT_MERGE_MS) to collect additional commits before translating. This prevents fragmented translations from aggressive VAD.
Stability Timeout
If VAD stalls (no new commits), a 3.5 second fallback timer (STABILITY_TIMEOUT_MS) fires to translate whatever new text has accumulated, preventing indefinite silence.
Text Validation
Before translation, text is validated against EN/RU/UK character regex patterns. This filters out hallucinated text from the STT model (common with silence or background noise).
Provider Chain
The system supports three translation providers with automatic fallback:
Default
LibreTranslate
Self-hosted, no API key required. Runs in Docker alongside the app. Best for privacy and cost.
Premium
DeepL
High-quality translations. Supports both free and paid API tiers. Auto-detects endpoint.
AI
Claude
Anthropic's Claude for context-aware translations. Uses claude-haiku-4-5 for speed.
Fallback Logic
1. Try primary provider (admin-selected)
2. If primary fails → try configured fallback
3. If fallback fails → try LibreTranslate (last resort)
4. If all fail → emit error event
Language Detection
The app uses a two-layer auto-detection approach:
Layer 1: Script-based Pre-detection
Before calling any translation API, the backend checks Unicode character scripts:
- Cyrillic characters (Unicode 0x0400–0x04FF) → if >50% of matched letters are Cyrillic, detected as Russian
- Latin characters → detected as English
- This avoids low-confidence results from LibreTranslate's
/detect endpoint on short text
Layer 2: STT Language Code
When the auto_language_detect flag is enabled, ElevenLabs Scribe returns a language_code with each transcript commit. The backend uses this to correctly route EN/RU/UK without relying solely on script detection.
Note: For LibreTranslate, both Russian and Ukrainian Cyrillic text is passed with source ru since LibreTranslate handles Ukrainian text acceptably via the Russian model. DeepL and Claude providers distinguish Ukrainian natively and handle uk as a proper source language.
Language Gating
Detected languages are checked against the admin-approved pool. If a detected language isn't in the allowed set, the translation is rejected to prevent hallucinated language outputs.
TTS Pipeline
After translation, the text is sent to ElevenLabs TTS:
const stream = await client.textToSpeech.stream(voiceId, {
text: translatedText,
model_id: "eleven_multilingual_v2",
output_format: "mp3_44100_128",
voice_settings: {
stability: 0.5,
similarity_boost: 0.75,
style: 0.0,
speed: 1.0,
use_speaker_boost: true
}
});
Audio Delivery
TTS audio is streamed to a Buffer, then emitted as a base64-encoded MP3 via the tts_audio Socket.io event.
Frontend Playback Queue
The frontend maintains an audio queue to prevent overlapping playback:
- Received
tts_audio events are queued
- Each segment plays to completion before the next starts
- A configurable pause (600ms default) is inserted between segments
- The pause duration is controlled by
tts_segment_pause_ms (adjustable in admin)
Microphone Input
- User selects "Mic" tab and chooses a TTS voice
- Browser captures audio via Web Audio API's
ScriptProcessor
- PCM 16-bit LE at 16kHz sample rate sent to backend via Socket.io
- Backend pipes audio to ElevenLabs Scribe v2 Realtime WebSocket
- Language auto-detected (EN/RU/UK), text translated and synthesized
- TTS audio returned and played back with inter-segment pauses
YouTube Input
- User pastes a YouTube URL (live stream or video)
- Backend spawns
yt-dlp | ffmpeg child processes
- Audio extracted as PCM stream (16kHz, 16-bit LE, mono)
- Piped to Scribe v2, same pipeline as microphone
- Stream ends when YouTube content ends or user stops
User Interface
The user view features a dark cavern theme with:
- Waveform visualizer — Canvas-based bar chart with orange gradient and cyan tips
- Transcript display — White translated text scrolls upward with fade masks
- Partial transcript — Shown in italic orange while STT is processing
- Source tabs — Toggle between Mic and YouTube (controlled by feature flags)
How It Works
The backend uses yt-dlp and ffmpeg as child processes to extract audio from YouTube URLs:
yt-dlp (best audio) → ffmpeg (PCM 16kHz 16-bit LE mono) → Scribe v2
Supported Sources
- Live streams — Translates in real-time as the stream progresses
- Regular videos — Processes the full audio track
- Any URL supported by yt-dlp (YouTube, etc.)
Requirements
Both yt-dlp and ffmpeg must be installed and available in the system PATH. On macOS:
brew install yt-dlp ffmpeg
⚠
Feature Flag Required
YouTube input is controlled by the youtube_input feature flag. Enable it in the admin panel to show the YouTube tab in the user view.
Overview
The Biblical Transcript Simulator is an admin-only feature that generates biblical text passages using Anthropic's Claude API, then routes them through the full translation pipeline. This provides a hands-free way to test STT → Translation → TTS without a live audio source.
Language Styles
| Language | Style | Example |
en |
King James English |
"In the beginning was the Word..." |
ru |
Church Slavonic Russian |
"В начале было Слово..." |
uk |
Traditional Ukrainian |
"На початку було Слово..." |
Flow
- Admin provides Anthropic API key and selects language
- Backend calls Claude with streaming (uses
claude-sonnet-4-6)
- Claude generates 6-8 biblical passages, 3-5 sentences each
- Stream is buffered until 140+ characters AND complete sentences
- Chunks emitted with 1800ms smooth pacing between them
- Each chunk flows through the standard pipeline:
- Emitted as
transcript (isFinal: true)
- Auto-translated via configured provider
- TTS synthesized and audio returned
- Frontend plays audio with standard inter-segment pause
💡
Feature Flag
Enable biblical_simulator in the admin feature flags panel. The Anthropic API key is provided at runtime in the UI — it's never stored in config files.
Overview
Voice Training uses ElevenLabs' Instant Voice Cloning (IVC) API to create custom voices from audio samples. Once cloned, the voice appears in the voice selector immediately.
From Microphone
- Open the Voice Training section in the admin panel
- Record multiple audio clips using your browser microphone
- Provide a name for the voice
- Clips are uploaded to ElevenLabs IVC API
- Cloned voice is available for TTS immediately
From YouTube
- Paste a YouTube URL in the Voice Training section
- Backend extracts N × 30-second clips via
yt-dlp + ffmpeg
- Clips are uploaded to ElevenLabs IVC API
- Resulting voice is stored in your ElevenLabs account
⚠
ElevenLabs Account
Cloned voices are stored in your ElevenLabs account, not locally. Ensure your plan supports voice cloning.
Concepts
| Concept | Description |
| Active Language Pair |
The current pair used for translation (e.g., EN ↔ RU, EN ↔ UK, or RU ↔ UK). Set by admin. |
| Available Languages |
The pool of languages viewers can select from (if user_language_selector is enabled). |
Admin Controls
- Change the active language pair via the admin panel
- Changes broadcast to all connected clients in real-time
- Manage the available languages pool for viewer selection
Viewer Selection
When the user_language_selector feature flag is enabled, viewers can override the admin-set language pair by selecting their own preferred languages from the available pool.
Feature Flags
Feature flags control runtime behavior and are stored in Redis with fallback defaults from application.yaml. All flags default to the values shown below but can be toggled live via the Admin Panel. Changes are broadcast to all connected clients in real time.
| Flag |
Default |
Description |
youtube_input |
true |
Enable YouTube audio stream input for speech-to-text. |
mic_input |
true |
Enable microphone audio input for speech-to-text. |
auto_language_detect |
true |
Automatically detect source language during transcription. |
user_language_selector |
false |
Allow viewers to select source & target language pair from available pool. |
audio_device_selector |
true |
Allow viewers to select their audio input device. |
Storage & Runtime Behavior
Feature flags are persisted in Redis with the prefix flag:. On startup, the backend merges Redis values with application.yaml defaults — Redis overrides always win. When a flag is toggled via the Admin API, it is immediately saved to Redis and broadcast to all connected WebSocket clients, ensuring real-time UI updates across all viewers.
Admin API
Feature flags can be managed via the admin endpoints. All requests require the X-Admin-Password header.
# Retrieve all feature flags (merged defaults + Redis overrides)
GET /admin/flags
X-Admin-Password: admin123
# Response:
{
"flags": {
"youtube_input": true,
"mic_input": true,
"auto_language_detect": true,
"user_language_selector": false,
"audio_device_selector": true
}
}
# Retrieve a single flag
GET /admin/flags/youtube_input
X-Admin-Password: admin123
# Response:
{
"flag": "youtube_input",
"value": true
}
# Toggle a flag
POST /admin/flags/youtube_input
X-Admin-Password: admin123
Content-Type: application/json
{
"value": false
}
# Response:
{
"flag": "youtube_input",
"value": false
}
WebSocket Events
When a flag is toggled via the admin API, the server broadcasts a feature_flags event to all connected WebSocket clients:
// Client receives merged flags on connect
socket.on('feature_flags', (flags) => {
console.log('Current flags:', flags);
// {
// youtube_input: true,
// mic_input: true,
// auto_language_detect: true,
// user_language_selector: false,
// audio_device_selector: true
// }
});
File Structure
| File | Purpose |
config/application.yaml |
Base defaults for all environments |
config/application-local.yaml |
Local development overrides (localhost URLs) |
config/application-prod.yaml |
Production overrides (Docker service names) |
The APP_ENV environment variable (local or prod) determines which overlay file is loaded on top of the base config.
Full Configuration Reference
server:
port: 3001
cors_origin: "http://localhost:5173"
elevenlabs:
api_key: "${ELEVENLABS_API_KEY}"
default_voice_id: "kxj9qk6u5PfI0ITgJwO0"
tts_model: "eleven_multilingual_v2"
tts_settings:
stability: 0.5
similarity_boost: 0.75
style: 0.0
speed: 1.0
use_speaker_boost: true
stt_model: "scribe_v2"
anthropic:
api_key: "${ANTHROPIC_API_KEY}"
deepl:
api_key: "${DEEPL_API_KEY}"
libretranslate:
url: "http://libretranslate:5000"
api_key: ""
redis:
host: "redis"
port: 6379
password: ""
feature_flags:
youtube_input: true
mic_input: true
auto_language_detect: true
user_language_selector: false
audio_device_selector: true
audio:
sample_rate: 16000
channels: 1
chunk_duration_ms: 250
translation:
source_lang: "auto"
target_lang_en: "en"
target_lang_ru: "ru"
provider: "libretranslate"
fallback: "libretranslate"
Environment Variable Interpolation
YAML values using ${VAR_NAME} syntax are automatically replaced with the corresponding environment variable at startup.
| Variable |
Required |
Default |
Description |
ELEVENLABS_API_KEY |
Yes |
— |
API key for ElevenLabs TTS & STT services. |
ELEVENLABS_VOICE_ID |
No |
kxj9qk6u5PfI0ITgJwO0 |
Default ElevenLabs voice ID for text-to-speech. |
ANTHROPIC_API_KEY |
No |
— |
API key for Anthropic Claude; used for sermon generation & Claude translation provider. Can also be set in Admin UI. |
DEEPL_API_KEY |
No |
— |
API key for DeepL translation provider. Leave empty if not using DeepL. |
TRANSLATION_PROVIDER |
No |
libretranslate |
Primary translation provider: deepl, claude, or libretranslate. Can be changed live in Admin UI. |
APP_ENV |
No |
local |
Application environment: local (development) or prod (Docker). |
FRONTEND_URL |
No |
http://localhost |
Frontend URL used for CORS origin in production; e.g., https://translate.example.com. |
LISTEN_PORT |
No |
80 |
Host port the frontend listens on. |
REDIS_PASSWORD |
No |
— |
Redis authentication password; leave empty for no auth. |
LIBRETRANSLATE_API_KEY |
No |
— |
Optional API key for LibreTranslate instance if it requires authentication. |
ADMIN_PASSWORD |
No |
admin123 |
Admin page protection password; must be changed in production. |
server.port |
No |
3001 |
Backend server port (from application.yaml). |
server.cors_origin |
No |
http://localhost:5173 |
CORS origin for frontend requests (from application.yaml). |
elevenlabs.tts_model |
No |
eleven_multilingual_v2 |
ElevenLabs TTS model ID (from application.yaml). |
elevenlabs.stt_model |
No |
scribe_v2 |
ElevenLabs STT model ID (from application.yaml). |
elevenlabs.tts_settings.stability |
No |
0.5 |
TTS voice stability parameter (from application.yaml). |
elevenlabs.tts_settings.similarity_boost |
No |
0.75 |
TTS similarity boost parameter (from application.yaml). |
elevenlabs.tts_settings.style |
No |
0.0 |
TTS style parameter (from application.yaml). |
elevenlabs.tts_settings.speed |
No |
1.0 |
TTS speech speed (from application.yaml). |
elevenlabs.tts_settings.use_speaker_boost |
No |
true |
Enable TTS speaker boost (from application.yaml). |
redis.host |
No |
redis |
Redis server hostname (from application.yaml). |
redis.port |
No |
6379 |
Redis server port (from application.yaml). |
libretranslate.url |
No |
http://libretranslate:5000 |
LibreTranslate service URL (from application.yaml). |
feature_flags.youtube_input |
No |
true |
Enable YouTube audio input source (from application.yaml). |
feature_flags.mic_input |
No |
true |
Enable microphone audio input source (from application.yaml). |
feature_flags.auto_language_detect |
No |
true |
Enable automatic language detection (from application.yaml). |
feature_flags.user_language_selector |
No |
false |
Allow viewers to select language pairs (from application.yaml). |
feature_flags.audio_device_selector |
No |
true |
Enable audio device selection interface (from application.yaml). |
audio.sample_rate |
No |
16000 |
Audio sample rate in Hz (from application.yaml). |
audio.channels |
No |
1 |
Number of audio channels (from application.yaml). |
audio.chunk_duration_ms |
No |
250 |
Audio chunk duration in milliseconds (from application.yaml). |
translation.source_lang |
No |
auto |
Source language for translation; auto enables auto-detection (from application.yaml). |
translation.target_lang_en |
No |
en |
Target language code for English streams (from application.yaml). |
translation.target_lang_ru |
No |
ru |
Target language code for Russian streams (from application.yaml). |
translation.provider |
No |
libretranslate |
Primary translation provider: deepl, claude, or libretranslate (from application.yaml). |
translation.fallback |
No |
libretranslate |
Fallback translation provider when primary fails (from application.yaml). |
TTS Settings
API Endpoints
GET /admin/tts-settings
Response: { "settings": { "stability": 0.5, "similarity_boost": 0.75, "style": 0.0, "speed": 1.0, "use_speaker_boost": true } }
POST /admin/tts-settings
Request: { "stability": 0.5, "similarity_boost": 0.75, "style": 0.0, "speed": 1.0, "use_speaker_boost": true }
Response: { "settings": { ... } }
| Setting |
Range |
Default |
Description |
stability |
0.0 – 1.0 |
0.5 |
Controls voice consistency; lower values increase variability, higher values are more stable. |
similarity_boost |
0.0 – 1.0 |
0.75 |
Emphasizes speaker similarity to the original voice; 0 = less similar, 1 = more similar. |
style |
0.0 – 1.0 |
0.0 |
Applies stylistic emphasis to speech; higher values add more exaggerated expression. |
speed |
0.5 – 2.0 |
1.0 |
Speech playback speed; 1.0 is normal, < 1.0 is slower, > 1.0 is faster. |
use_speaker_boost |
true | false |
true |
Applies speaker boost technology to enhance voice clarity and presence. |
ElevenLabs Configuration (from application.yaml)
| Configuration |
Value |
Description |
tts_model |
eleven_multilingual_v2 |
ElevenLabs multilingual TTS model ID for speech synthesis. |
stt_model |
scribe_v2 |
ElevenLabs Scribe v2 model for speech-to-text transcription (supports VAD & commit-based finalization). |
default_voice_id |
kxj9qk6u5PfI0ITgJwO0 |
Default voice ID used when no voice is explicitly selected. |
STT Timing Settings
API Endpoints
GET /admin/stt-timing
Response: { "settings": { "commit_merge_ms": 2500, "stability_timeout_ms": 3500, "tts_segment_pause_ms": 600 } }
POST /admin/stt-timing
Request: { "commit_merge_ms": 2500, "stability_timeout_ms": 3500, "tts_segment_pause_ms": 600 }
Response: { "settings": { ... } }
| Setting |
Range |
Default |
Description |
commit_merge_ms |
0 – 10000 |
2500 |
Time in milliseconds to buffer VAD commits before translating; merges sentence fragments into coherent units. |
stability_timeout_ms |
0 – 10000 |
3500 |
Time in milliseconds to wait for stable partial text before translating when VAD does not commit. |
tts_segment_pause_ms |
0 – 2000 |
600 |
Pause in milliseconds between TTS audio segments; frontend uses this for playback timing. |
Notes
- TTS settings are persisted in Redis and loaded at startup; changes via
POST /admin/tts-settings take effect immediately for subsequent synthesis requests.
- STT timing settings control how speech-to-text commits and stability detection work;
commit_merge_ms buffers VAD-fired commits to avoid translating mid-sentence fragments.
- All settings are admin-protected via
X-Admin-Password header (default: admin123—change in production via ADMIN_PASSWORD environment variable).
STT Timing Settings
Configure Speech-to-Text recognition timing & translation triggers.
| Setting |
Default |
Description |
commit_merge_ms |
2500 |
Buffer VAD commits for this duration before translating; merges sentence fragments (ms). |
stability_timeout_ms |
3500 |
Timeout for stable partial text fallback if VAD never commits (ms). |
tts_segment_pause_ms |
600 |
Pause duration between consecutive TTS audio segment playbacks (ms); frontend-only. |
Tuning Guide
- Slow speakers or long pauses: Increase
commit_merge_ms (e.g., 3500–4500) to avoid splitting mid-sentence; increase stability_timeout_ms proportionally.
- Fast speakers or rapid fragments: Decrease
commit_merge_ms (e.g., 1500–2000) to translate sooner; lower stability_timeout_ms if VAD is responsive.
- TTS playback too fast or choppy: Increase
tts_segment_pause_ms to 800–1000 for smoother transitions between audio clips.
- Silent detection: VAD (Voice Activity Detection) auto-commits on silence — both timers serve as fallbacks when VAD is unreliable.
API
GET /admin/stt-timing → Retrieve current STT timing settings.
curl -X GET http://localhost:3001/admin/stt-timing \
-H "x-admin-password: admin123"
POST /admin/stt-timing → Update one or more STT timing settings.
curl -X POST http://localhost:3001/admin/stt-timing \
-H "x-admin-password: admin123" \
-H "Content-Type: application/json" \
-d '{
"commit_merge_ms": 3000,
"stability_timeout_ms": 4000,
"tts_segment_pause_ms": 700
}'
Response (both GET & POST):
{
"settings": {
"commit_merge_ms": 2500,
"stability_timeout_ms": 3500,
"tts_segment_pause_ms": 600
}
}
Notes
- Settings are persisted to Redis and survive server restarts.
- Changes take effect immediately for new sessions; active sessions use the previous settings.
tts_segment_pause_ms is broadcast to all connected clients via the stt_timing socket event.
- Commit buffer flushes after
commit_merge_ms of silence — useful for merging VAD-fragmented utterances like "Today we are going to" + "talk about something".
- Stability timeout fires if partial text is unchanged for
stability_timeout_ms — fallback for slow or unresponsive VAD.
Authentication: All endpoints require x-admin-password header matching ADMIN_PASSWORD environment variable (default: admin123).
API Keys
Retrieve the stored Anthropic API key.
Get status of all configured API keys (elevenlabs, anthropic, deepl, libretranslate).
Update one or more API keys.
Body: {
"elevenlabs": "string?",
"anthropic": "string?",
"deepl": "string?",
"libretranslate": "string?"
}
Voice Management
Scan and list all available ElevenLabs voices with detailed logging of new voices.
Get the list of voice IDs allowed for viewers to select from.
Set the whitelist of allowed voice IDs — broadcasts to all connected clients.
Body: {
"voiceIds": ["string", "string"]
}
Feature Flags
Get all feature flags merged from YAML defaults & Redis overrides.
Get a single feature flag value.
Set a feature flag value — broadcasts to all connected clients via WebSocket.
Body: {
"value": boolean
}
TTS & STT Settings
Get current TTS voice settings (stability, similarity_boost, style, speed, use_speaker_boost).
Update TTS voice settings.
Body: {
"stability": "number?",
"similarity_boost": "number?",
"style": "number?",
"speed": "number?",
"use_speaker_boost": "boolean?"
}
Get STT timing settings (commit_merge_ms, stability_timeout_ms, tts_segment_pause_ms).
Update STT timing configuration.
Body: {
"commit_merge_ms": "number?",
"stability_timeout_ms": "number?",
"tts_segment_pause_ms": "number?"
}
Languages
Get the currently active language pair [source, target].
Set the active language pair — must be exactly 2 language codes.
Body: {
"languages": ["en", "ru"]
}
Get the pool of languages available for viewers to select from.
Set the pool of allowed languages — broadcasts to all connected clients.
Body: {
"languages": ["en", "ru", "uk"]
}
Translation Provider
Get the active translation provider & list of available options (deepl, claude, libretranslate).
Set the active translation provider.
Body: {
"provider": "deepl" | "claude" | "libretranslate"
}
Audio Device
Get the admin-selected audio input device that overrides viewer selection.
Set the forced audio input device — broadcasts to all connected clients.
Body: {
"deviceId": "string?",
"label": "string?"
}
Content Generation
Generate a biblical sermon snippet via Claude — language specified as en, ru, or uk.
Body: {
"apiKey": "string?",
"language": "en" | "ru" | "uk"
}
Voice Training
Clone a voice from browser microphone recordings (base64-encoded audio blobs).
Body: {
"name": "string",
"clips": ["base64_string", "base64_string"],
"mimeType": "audio/webm?"
}
Clone a voice from YouTube URL — extracts N×30s clips via yt-dlp & ffmpeg then uploads.
Body: {
"name": "string",
"youtubeUrl": "string",
"clipCount": "number?",
"startOffset": "number?"
}
Socket.io Events
Server → Client Events
| Event |
Payload |
Description |
feature_flags |
Record<string, boolean> |
Merged feature flags from config defaults & Redis overrides on connection or admin update. |
languages |
{ languages: string[] } |
Current active language pair (2-element array); broadcast when admin or viewer changes languages. |
available_languages |
{ languages: string[] } |
Pool of languages available for viewer selection; broadcast when admin updates the pool. |
available_voices |
{ voiceIds: string[] } |
Pool of ElevenLabs voice IDs available for selection; broadcast when admin updates allowed voices. |
stt_timing |
{ tts_segment_pause_ms: number } |
STT timing configuration including pause between TTS audio segments. |
admin_audio_device |
{ deviceId: string; label: string } |
Admin-selected audio input device that overrides viewer’s local selection. |
session_started |
{ source: 'mic' | 'youtube' | 'biblical' } |
Translation session initiated with specified audio source. |
transcript |
{ text: string; isFinal: boolean } |
Speech-to-text output from ElevenLabs Scribe or Biblical simulator. |
translation |
{ original: string; translated: string; detectedLanguage: string } |
Translated text & detected source language after processing final transcript. |
tts_audio |
{ audio: string } |
Base64-encoded MP3 audio buffer of translated text generated via ElevenLabs TTS. |
audio_level |
{ data: number[] } |
Waveform data (64 normalized samples) for real-time audio level visualization. |
error |
{ message: string } |
Error message from speech recognition, translation, TTS, or stream processing. |
stream_ended |
{} |
YouTube or Biblical stream completed or stopped. |
session_stopped |
{} |
Translation session stopped by client or server. |
admin_translate_result |
{ original: string; translated: string; detectedLanguage: string; audio: string } |
Result of admin instant translate & TTS test with custom language pair. |
Client → Server Events
| Event |
Payload |
Description |
set_languages |
{ languages: string[] } |
Viewer selects a 2-language pair from the available pool; broadcasts update to all clients. |
start_session |
{ voiceId?: string; source: 'mic' | 'youtube'; youtubeUrl?: string } |
Begin STT session from microphone or YouTube audio stream with optional TTS voice override. |
audio_chunk |
{ audio: string } |
Base64-encoded PCM audio chunk from browser microphone; always emits waveform to client. |
stop_session |
{} |
Stop active STT session and cleanup resources. |
admin_translate_test |
{ text: string; voiceId?: string; sourceLang?: string; targetLang?: string } |
Admin instant translate & TTS test bypassing viewer language-pair restriction. |
start_biblical_sim |
{ anthropicApiKey: string; language: 'en' | 'ru' | 'uk'; voiceId?: string } |
Start Biblical text simulator using Anthropic to generate sermon excerpts with optional TTS voice. |
stop_biblical_sim |
{} |
Stop active Biblical text simulator stream. |
SDK
Uses the official @elevenlabs/elevenlabs-js SDK (v2). The client is lazy-loaded on first use.
Speech-to-Text (Scribe v2 Realtime)
Connects via native WebSocket to wss://api.elevenlabs.io/v1/speech-to-text/realtime. Handles:
- VAD-based commit buffering with configurable merge window
- Stability timeout fallback for stalled VAD
- Text validation (EN/RU/UK character regex filtering)
- Partial and final transcript emission
Text-to-Speech
Uses client.textToSpeech.stream() with the eleven_multilingual_v2 model. Audio is collected into a Buffer and emitted as base64 MP3.
Voice Management
client.voices.getAll() — fetches all voices from account
- Admin can filter which voices are available to viewers
- Voice cloning via IVC API (from recordings or YouTube)
Key File
backend/src/services/elevenlabs.service.ts
Provider Details
LibreTranslate
Self-hosted in Docker. No API key required by default. Provides language detection and translation via REST API.
File: backend/src/services/libretranslate.service.ts
DeepL
Premium translation API. Auto-detects free vs. paid endpoint based on the API key format.
File: backend/src/services/deepl.service.ts
Claude (Anthropic)
AI-powered translation using claude-haiku-4-5 for speed. Includes language detection and auto-flip logic.
File: backend/src/services/claude-translate.service.ts
Routing
Provider routing is handled by backend/src/services/translation.provider.ts:
- Try admin-selected primary provider
- On failure, try configured fallback provider
- LibreTranslate is always the last-resort fallback
Connection
Uses ioredis with automatic retry strategy. Falls back to in-memory/YAML defaults if Redis is unavailable.
Key Patterns
| Pattern | Example | Purpose |
flag:<name> |
flag:youtube_input |
Feature flag boolean values |
setting:<name> |
setting:tts_settings |
JSON settings objects |
Key File
backend/src/services/redis.service.ts
Local Development
Use docker-compose.local.yml for Redis and LibreTranslate only (backend/frontend run natively):
docker compose -f docker-compose.local.yml up -d
Production
Use docker-compose.yml for all services:
docker compose up -d --build
Services
| Service | Image | Port | Notes |
| frontend |
Nginx (custom build) |
80 (exposed) |
Serves React build, proxies API/WS to backend |
| backend |
Node.js (custom build) |
3001 (internal) |
Express + Socket.io server |
| redis |
redis:7-alpine |
6379 (internal) |
Feature flags and settings store |
| libretranslate |
libretranslate/libretranslate |
5000 (internal) |
Self-hosted translation engine |
Configuration
ELEVENLABS_API_KEY=sk-your-production-key
ADMIN_PASSWORD=strong-secure-password
FRONTEND_URL=https://translate.example.com
APP_ENV=prod
REDIS_PASSWORD=redis-secret
Deploy
docker compose up -d --build
Reverse Proxy
When running behind Nginx or another reverse proxy:
- Set
LISTEN_PORT in .env (e.g., 8080)
- Proxy pass to
localhost:8080
- Important: Ensure WebSocket upgrades are forwarded for the
/socket.io/ path
server {
listen 443 ssl;
server_name translate.example.com;
location / {
proxy_pass http://localhost:8080;
proxy_http_version 1.1;
proxy_set_header Upgrade $http_upgrade;
proxy_set_header Connection "upgrade";
proxy_set_header Host $host;
}
}
Monitoring
# Check all services
docker compose ps
# View backend logs
docker compose logs -f backend
# Health check
curl http://localhost:3001/api/health
Shipped
v0.1 – v0.2 — Core Translation Engine
- Real-time STT via ElevenLabs Scribe v2 Realtime
- Multi-provider translation (LibreTranslate, DeepL, Claude)
- TTS voice synthesis with ElevenLabs
- Microphone and YouTube live input
- Admin panel with feature flags, voice management, TTS tuning
- Biblical Transcript Simulator for pipeline testing
- Instant Voice Cloning from recordings and YouTube
Shipped
v0.3 — Audio Mixer & Device Selection
Browser-side audio device scanning with support for professional mixing consoles, virtual audio devices, and audio interfaces.
- Browser-side device enumeration with permission flow
- Virtual device detection (Loopback, BlackHole, VB-Audio, Voicemeeter, OBS)
- Categorized device picker (Microphones vs Mixers / Virtual Devices)
- Admin device override broadcast to all viewers via Socket.io
- Real-time feature flag broadcasting
Up Next
v0.4 — Direct Audio Interface Feed
Accept audio directly from professional mixing consoles and audio interfaces — bypass browser mic capture entirely for broadcast-quality input.
- Direct audio interface input (ASIO / Core Audio / ALSA)
- Multi-channel mixer feed support
- Low-latency audio routing (sub-100ms)
- Hardware device auto-discovery and selection
- Professional broadcast integration (NDI, Dante)
Up Next
v0.5 — Video Translation
Full video translation pipeline — not just audio. Translate video content with synchronized subtitles and dubbed audio output.
- Video file upload and URL ingestion
- Synchronized subtitle generation (SRT/VTT)
- Dubbed audio track with voice-matched TTS
- Video player with real-time translated overlay
- Batch video processing queue
Planned
Future
- Additional language pairs beyond EN/RU/UK
- Speaker diarization (multi-speaker detection)
- Translation memory and glossary support
- Webhooks and API for third-party integrations
- Multi-tenant deployment with user accounts