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
Script-based Language Detection
LibreTranslate's /detect endpoint returns 0-confidence for short Cyrillic phrases. The app checks Unicode character scripts first (0x0400-0x04FF = Cyrillic) before calling the API.
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
Before translation, the pipeline performs script-based language pre-detection:
- Cyrillic characters (Unicode 0x0400-0x04FF) → detected as Russian or Ukrainian
- Latin characters → detected as English
- This avoids low-confidence results from LibreTranslate's
/detect endpoint on short text
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 are defined in config/application.yaml with built-in defaults. At runtime, Redis overrides take precedence — allowing the admin panel to toggle flags without restarting the server. Clients receive merged flags (YAML defaults + Redis overrides) on connection, and updates broadcast via Socket.IO.
| Flag |
Default |
Description |
youtube_input |
true |
Enable audio input from YouTube URLs via yt-dlp extraction. |
mic_input |
true |
Enable real-time speech input from the browser microphone. |
auto_language_detect |
true |
Automatically detect source language via ElevenLabs Scribe (leave language_code blank). |
user_language_selector |
false |
Allow viewers to pick language pairs from the admin-curated available languages pool. |
audio_device_selector |
true |
Allow viewers to select their audio input device (microphone); admin can force override. |
Runtime API
Feature flags are toggled via the admin REST API and persisted in Redis:
# Get all flags (merged YAML + Redis)
GET /admin/flags
→ { "flags": { "youtube_input": true, "mic_input": true, ... } }
# Get single flag
GET /admin/flags/:flag
→ { "flag": "youtube_input", "value": true }
# Set flag (persists to Redis, broadcasts to all clients)
POST /admin/flags/:flag
Content-Type: application/json
{ "value": false }
→ { "flag": "youtube_input", "value": false }
# Broadcasts: socket.io emit('feature_flags', merged)
Client Reception
On Socket.IO connection, the server emits merged flags to each client:
socket.on('feature_flags', (flags: Record<string, boolean>) => {
// flags = { youtube_input: true, mic_input: true, ... }
// Use to show/hide UI elements, enable/disable features, etc.
});
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_realtime"
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: false
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 text-to-speech & speech-to-text services. |
ELEVENLABS_VOICE_ID |
No |
kxj9qk6u5PfI0ITgJwO0 |
Default voice ID for TTS output when no voice is selected. |
ANTHROPIC_API_KEY |
No |
— |
API key for Anthropic Claude — used for sermon generation & Claude translation provider; can be set in Admin UI at runtime. |
DEEPL_API_KEY |
No |
— |
API key for DeepL translation provider (free tier ends with :fx); leave empty if not using DeepL. |
TRANSLATION_PROVIDER |
No |
libretranslate |
Primary translation provider: deepl | claude | 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 — set to actual domain (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 authentication. |
LIBRETRANSLATE_API_KEY |
No |
— |
API key for LibreTranslate instance if authentication is required; leave empty if not needed. |
ADMIN_PASSWORD |
No |
admin123 |
Admin page protection password — must be changed in production. |
TTS Settings
Configure text-to-speech parameters for ElevenLabs voice synthesis.
API Endpoints
GET /admin/tts-settings
Returns current TTS settings object.
POST /admin/tts-settings
Updates TTS settings. Request body: Partial<TtsSettings>
Returns updated settings object.
Settings Reference
| Setting |
Range |
Default |
Description |
stability |
0.0 – 1.0 |
0.5 |
Controls voice stability; lower values allow more variation, higher values are more consistent. |
similarity_boost |
0.0 – 1.0 |
0.75 |
Boosts similarity to the original voice model; higher values increase adherence to the voice profile. |
style |
0.0 – 1.0 |
0.0 |
Exaggerates speech style; 0.0 is neutral, higher values add expressiveness. |
speed |
0.5 – 2.0 |
1.0 |
Speech rate multiplier; 1.0 is normal speed, <1.0 is slower, >1.0 is faster. |
use_speaker_boost |
true — false |
true |
Applies speaker boost optimization to enhance voice clarity and presence. |
STT Timing Settings
Fine-tune speech-to-text processing timing for commit buffering and stability detection.
GET /admin/stt-timing
Returns current STT timing settings object.
POST /admin/stt-timing
Updates STT timing settings. Request body: Partial<SttTimingSettings>
Returns updated settings object.
| Setting |
Range |
Default |
Description |
commit_merge_ms |
100 – 10000 ms |
2500 |
Duration to buffer VAD-triggered commits before translating; merges short fragments into coherent segments. |
stability_timeout_ms |
100 – 10000 ms |
3500 |
Time to wait for partial text to stabilize before translating if no VAD commit fires. |
tts_segment_pause_ms |
100 – 2000 ms |
600 |
Pause duration between consecutive TTS audio segments on the frontend; controls playback pacing. |
Configuration File Defaults
TTS settings are initialized from config/application.yaml under the elevenlabs.tts_settings key and can be overridden at runtime via the admin API. Settings are persisted to Redis.
STT Timing Settings
Configure speech-to-text recognition and translation pipeline timing parameters. These settings control how long the system waits before processing recognized speech segments.
Settings Reference
| Setting |
Default |
Description |
commit_merge_ms |
2500 |
Buffer VAD commits for this duration before translating; merges short fragments into complete thoughts (ms). |
stability_timeout_ms |
3500 |
Trigger translation when partial text remains unchanged for this duration; acts as fallback when VAD is slow (ms). |
tts_segment_pause_ms |
600 |
Pause between TTS audio segment playback on the frontend; prevents clipped transitions (ms). |
Tuning Guide
- For faster responsiveness: Reduce
commit_merge_ms and stability_timeout_ms to 1000–1500ms. Trade-off: incomplete sentences may trigger premature translation.
- For natural sentence bundling: Increase
commit_merge_ms to 3000–4000ms. Waits longer for speaker pauses to complete thoughts.
- For smooth audio playback: Adjust
tts_segment_pause_ms between 400–800ms. Lower values play segments faster; higher values allow time for buffer refills.
- VAD behavior: If VAD (Voice Activity Detection) is firing erratically, stability timeout acts as a safety net. Increase both timings to reduce false triggers.
- Hallucination filtering: All recognized text is validated against English, Russian, and Ukrainian character sets. Invalid scripts are silently dropped during processing.
API Endpoints
GET /admin/stt-timing
Retrieve current STT timing configuration.
curl -X GET http://localhost:3001/admin/stt-timing \
-H "x-admin-password: admin123"
{
"settings": {
"commit_merge_ms": 2500,
"stability_timeout_ms": 3500,
"tts_segment_pause_ms": 600
}
}
POST /admin/stt-timing
Update one or more STT timing settings. Unspecified fields retain current values.
curl -X POST http://localhost:3001/admin/stt-timing \
-H "x-admin-password: admin123" \
-H "Content-Type: application/json" \
-d '{
"commit_merge_ms": 3000,
"tts_segment_pause_ms": 700
}'
{
"settings": {
"commit_merge_ms": 3000,
"stability_timeout_ms": 3500,
"tts_segment_pause_ms": 700
}
}
Real-Time Socket Emission
Connected frontend clients receive STT timing updates via Socket.IO on connection and after any admin configuration change:
socket.on('stt_timing', (data) => {
console.log('TTS segment pause (ms):', data.tts_segment_pause_ms);
});
Persistence
All STT timing settings are persisted in Redis under the key setting:stt_timing. Changes survive server restart. Initial values are loaded from the configuration file at startup and overridden by Redis on each connection.
Authentication: All endpoints require x-admin-password header matching ADMIN_PASSWORD environment variable (default: admin123).
API Keys
Retrieve 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"
}
Get the currently configured Anthropic API key.
Voice Management
Scan and list all available ElevenLabs voices, marking new additions not yet in the allowed list.
Get the list of voice IDs available for viewers to select from (admin-curated whitelist).
Set the whitelist of voice IDs viewers can choose from; broadcasts update to all connected clients.
Body: {
"voiceIds": ["voice_id_1", "voice_id_2"]
}
Feature Flags
Retrieve all feature flags merged from YAML config defaults and Redis overrides.
Get the value of a specific feature flag by name.
Set a feature flag value and broadcast the update to all connected WebSocket clients.
TTS & STT Settings
Get current TTS settings (stability, similarity_boost, style, speed, use_speaker_boost).
Update TTS voice settings; partial updates are merged with current values.
Body: {
"stability": 0.5,
"similarity_boost": 0.75,
"style": 0.0,
"speed": 1.0,
"use_speaker_boost": true
}
Get STT timing settings (VAD commit buffer merge delay, stability timeout, TTS segment pause).
Update STT timing thresholds controlling when transcripts are translated.
Body: {
"commit_merge_ms": 2500,
"stability_timeout_ms": 3500,
"tts_segment_pause_ms": 600
}
Languages
Get the currently active language pair [source, target].
Set the active language pair and broadcast to all connected clients.
Body: {
"languages": ["en", "ru"]
}
Get the pool of language codes available for viewers to select from.
Set the language pool and broadcast both the pool and current active pair to all clients.
Body: {
"languages": ["en", "ru", "uk"]
}
Translation Provider
Get the currently active translation provider (deepl, claude, or libretranslate) and list available options.
Switch the active translation provider.
Body: {
"provider": "libretranslate"
}
Audio Device
Get the admin-selected audio input device ID and label that overrides viewer selection.
Force a specific audio input device for all viewers; broadcasts to all connected clients.
Body: {
"deviceId": "device_id_string",
"label": "Microphone Name"
}
Voice Training & Cloning
Clone a voice from base64-encoded browser microphone recordings; uploads to ElevenLabs Instant Voice Cloning.
Body: {
"name": "My Cloned Voice",
"clips": ["base64_audio_chunk_1", "base64_audio_chunk_2"],
"mimeType": "audio/webm"
}
Clone a voice by extracting N×30s clips from a YouTube video using yt-dlp and ffmpeg.
Body: {
"name": "YouTube Voice",
"youtubeUrl": "https://www.youtube.com/watch?v=...",
"clipCount": 3,
"startOffset": 0
}
Content Generation
Generate a 1–2 sentence biblical sermon excerpt via Claude Haiku in the specified language.
Body: {
"apiKey": "sk-ant-...",
"language": "ru"
}
Socket.io Events
Server → Client Events
| Event |
Payload |
Description |
feature_flags |
{ [flag: string]: boolean } |
Merged feature flags from YAML defaults & Redis overrides on connection & updates. |
languages |
{ languages: string[] } |
Current active language pair (source & target codes) on connection & when changed. |
available_languages |
{ languages: string[] } |
Pool of languages admin has approved; viewers pick from this list on connection & updates. |
available_voices |
{ voiceIds: string[] } |
Pool of voice IDs admin has approved; viewers pick from this list when updated. |
stt_timing |
{ tts_segment_pause_ms: number } |
Pause duration (ms) between TTS audio segments sent to frontend on connection. |
admin_audio_device |
{ deviceId: string; label: string } |
Admin-forced audio input device; viewer client uses this instead of local selection on connection & updates. |
session_started |
{ source: 'mic' | 'youtube' | 'biblical' } |
Session has begun for the specified source type. |
transcript |
{ text: string; isFinal: boolean } |
STT partial or final transcript text received from speech recognition or biblical simulator. |
translation |
{ original: string; translated: string; detectedLanguage: string } |
Final translated text with detected source language after processing a transcript. |
tts_audio |
{ audio: string } |
Base64-encoded MP3 audio chunk of translated text (TTS output). |
audio_level |
{ data: number[] } |
Waveform visualization data (64 downsampled amplitude samples) for audio level bars. |
stream_ended |
{} |
YouTube or biblical stream has ended; session should stop. |
session_stopped |
{} |
Session has been stopped (mic/YouTube/biblical ended or user clicked stop). |
admin_translate_result |
{ original: string; translated: string; detectedLanguage: string; audio: string } |
Result of admin instant translate & TTS test with base64-encoded MP3 audio. |
error |
{ message: string } |
Error message from STT, translation, TTS, or stream processing. |
Client → Server Events
| Event |
Payload |
Description |
set_languages |
{ languages: string[] } |
Viewer selects a new language pair (validated against available pool & broadcast to all clients). |
start_session |
{ voiceId?: string; source: 'mic' | 'youtube'; youtubeUrl?: string } |
Start STT session from microphone or YouTube URL with optional voice ID override. |
audio_chunk |
{ audio: string } |
Base64-encoded PCM audio chunk from browser mic; always triggers waveform emit. |
stop_session |
{} |
Stop the active mic or YouTube session. |
admin_translate_test |
{ text: string; voiceId?: string; sourceLang?: string; targetLang?: string } |
Admin instant translate & TTS without language-pair restriction; returns translation + audio. |
start_biblical_sim |
{ anthropicApiKey: string; language: 'en' | 'ru' | 'uk'; voiceId?: string } |
Start biblical text simulator using Anthropic API; streams sentences with auto-translation & TTS. |
stop_biblical_sim |
{} |
Stop the running 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