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# voicebox Backend
Production-quality FastAPI backend for Qwen3-TTS voice cloning.
## Features
-**Voice Profile Management** - Create, update, delete voice profiles with multi-sample support
-**Voice Cloning** - Generate speech using voice profiles with caching
-**Generation History** - Full history tracking with search and filtering
-**Transcription** - Whisper-based audio transcription
-**Multi-Sample Profiles** - Combine multiple reference samples for better quality
-**Voice Prompt Caching** - Dual memory + disk caching for fast generation
-**Audio Validation** - Automatic validation of reference audio quality
-**Model Management** - Lazy loading and VRAM management
## Architecture
```
backend/
├── main.py # FastAPI app with all routes
├── models.py # Pydantic request/response models
├── platform_detect.py # Platform detection for backend selection
├── tts.py # TTS backend abstraction (delegates to MLX or PyTorch)
├── transcribe.py # STT backend abstraction (delegates to MLX or PyTorch)
├── backends/ # Backend implementations
│ ├── __init__.py # Backend factory and protocols
│ ├── mlx_backend.py # MLX backend (Apple Silicon)
│ └── pytorch_backend.py # PyTorch backend (Windows/Linux/Intel)
├── profiles.py # Voice profile CRUD
├── history.py # Generation history
├── studio.py # Audio editing (TODO)
├── database.py # SQLite ORM
└── utils/
├── audio.py # Audio processing utilities
├── cache.py # Voice prompt caching
└── validation.py # Input validation
```
### Backend Selection
Voicebox automatically selects the best backend based on platform:
- **Apple Silicon (M1/M2/M3)**: Uses MLX backend with native Metal acceleration (4-5x faster)
- **Windows/Linux/Intel Mac**: Uses PyTorch backend (CUDA GPU if available, CPU fallback)
The backend is detected at runtime via `platform_detect.py`. Both backends implement the same interface, so the API remains consistent across platforms.
## API Endpoints
### Health & Info
#### `GET /`
Root endpoint with version info.
#### `GET /health`
Health check with model status.
**Response:**
```json
{
"status": "healthy",
"model_loaded": true,
"gpu_available": true,
"gpu_type": "Metal (Apple Silicon via MLX)",
"backend_type": "mlx",
"vram_used_mb": null
}
```
**Backend Types:**
- `"mlx"` - MLX backend (Apple Silicon with Metal acceleration)
- `"pytorch"` - PyTorch backend (Windows/Linux/Intel Mac)
### Voice Profiles
**Note:** The database is automatically initialized when the server starts. No manual setup required.
#### `POST /profiles`
Create a new voice profile.
**Request:**
```json
{
"name": "My Voice",
"description": "Optional description",
"language": "en"
}
```
**Response:**
```json
{
"id": "uuid",
"name": "My Voice",
"description": "Optional description",
"language": "en",
"created_at": "2024-01-01T00:00:00Z",
"updated_at": "2024-01-01T00:00:00Z"
}
```
#### `GET /profiles`
List all voice profiles.
#### `GET /profiles/{profile_id}`
Get a specific profile.
#### `PUT /profiles/{profile_id}`
Update a profile.
#### `DELETE /profiles/{profile_id}`
Delete a profile and all associated samples.
#### `POST /profiles/{profile_id}/samples`
Add a sample to a profile.
**Form Data:**
- `file`: Audio file (WAV, MP3, etc.)
- `reference_text`: Transcript of the audio
**Response:**
```json
{
"id": "sample-uuid",
"profile_id": "profile-uuid",
"audio_path": "/path/to/sample.wav",
"reference_text": "This is my voice"
}
```
#### `GET /profiles/{profile_id}/samples`
List all samples for a profile.
#### `DELETE /profiles/samples/{sample_id}`
Delete a specific sample.
### Generation
#### `POST /generate`
Generate speech from text using a voice profile.
**Request:**
```json
{
"profile_id": "uuid",
"text": "Hello, this is a test.",
"language": "en",
"seed": 42
}
```
**Response:**
```json
{
"id": "generation-uuid",
"profile_id": "profile-uuid",
"text": "Hello, this is a test.",
"language": "en",
"audio_path": "/path/to/audio.wav",
"duration": 2.5,
"seed": 42,
"created_at": "2024-01-01T00:00:00Z"
}
```
### History
#### `GET /history`
List generation history with optional filters.
**Query Parameters:**
- `profile_id` (optional): Filter by profile
- `search` (optional): Search in text content
- `limit` (default: 50): Results per page
- `offset` (default: 0): Pagination offset
#### `GET /history/{generation_id}`
Get a specific generation.
#### `DELETE /history/{generation_id}`
Delete a generation.
#### `GET /history/stats`
Get generation statistics.
**Response:**
```json
{
"total_generations": 100,
"total_duration_seconds": 250.5,
"generations_by_profile": {
"profile-uuid-1": 50,
"profile-uuid-2": 50
}
}
```
### Audio Files
#### `GET /audio/{generation_id}`
Download generated audio file.
Returns WAV file with appropriate headers.
### Transcription
#### `POST /transcribe`
Transcribe audio file to text.
**Form Data:**
- `file`: Audio file
- `language` (optional): Language hint (en or zh)
**Response:**
```json
{
"text": "Transcribed text here",
"duration": 5.5
}
```
### Model Management
#### `POST /models/load`
Manually load TTS model.
**Query Parameters:**
- `model_size`: Model size (1.7B or 0.6B)
#### `POST /models/unload`
Unload TTS model to free memory.
## Database Schema
### profiles
- `id`: UUID primary key
- `name`: Profile name (unique)
- `description`: Optional description
- `language`: Language code (en/zh)
- `created_at`: Creation timestamp
- `updated_at`: Last update timestamp
### profile_samples
- `id`: UUID primary key
- `profile_id`: Foreign key to profiles
- `audio_path`: Path to audio file
- `reference_text`: Transcript
### generations
- `id`: UUID primary key
- `profile_id`: Foreign key to profiles
- `text`: Generated text
- `language`: Language code
- `audio_path`: Path to audio file
- `duration`: Duration in seconds
- `seed`: Random seed (optional)
- `created_at`: Creation timestamp
### projects
- `id`: UUID primary key
- `name`: Project name
- `data`: JSON data
- `created_at`: Creation timestamp
- `updated_at`: Last update timestamp
## File Structure
```
data/
├── profiles/
│ └── {profile_id}/
│ ├── {sample_id}.wav
│ └── ...
├── generations/
│ └── {generation_id}.wav
├── cache/
│ └── {hash}.prompt
├── projects/
│ └── {project_id}.json
└── voicebox.db
```
## Setup
### 1. Install Dependencies
```bash
pip install -r requirements.txt
```
**Note:** On Apple Silicon, also install MLX dependencies for faster inference:
```bash
pip install -r requirements-mlx.txt
```
### 2. Download Models (Automatic)
The Qwen3-TTS models are automatically downloaded from HuggingFace Hub on first use, similar to how Whisper models work.
**No manual download required!** The models will be cached locally after the first download.
Available models:
- **1.7B** (recommended): `Qwen/Qwen3-TTS-12Hz-1.7B-Base` (~4GB)
- **0.6B** (faster): `Qwen/Qwen3-TTS-12Hz-0.6B-Base` (~2GB)
**Note:** The first generation will take longer as the model downloads. Subsequent generations will use the cached model.
#### Manual Download (Optional)
If you prefer to download models manually or have limited internet during runtime:
```bash
# Install huggingface-cli
pip install huggingface_hub
# Download 1.7B model
huggingface-cli download Qwen/Qwen3-TTS-12Hz-1.7B-Base
# Or use Python
python -c "from huggingface_hub import snapshot_download; snapshot_download('Qwen/Qwen3-TTS-12Hz-1.7B-Base')"
```
Models are cached in `~/.cache/huggingface/hub/` by default.
### 4. Run Server
```bash
# Development (local only)
python -m backend.main
# Production (allow remote access)
python -m backend.main --host 0.0.0.0 --port 8000
```
## Usage Examples
The desktop app, web client, and current development workflow use `http://localhost:17493` by default.
If you launch the backend manually with a different host or port, substitute that address in the examples below.
### Creating a Voice Profile
```bash
# 1. Create profile
curl -X POST http://localhost:17493/profiles \
-H "Content-Type: application/json" \
-d '{"name": "My Voice", "language": "en"}'
# Response: {"id": "abc-123", ...}
# 2. Add sample
curl -X POST http://localhost:17493/profiles/abc-123/samples \
-F "[email protected]" \
-F "reference_text=This is my voice sample"
```
### Generating Speech
```bash
curl -X POST http://localhost:17493/generate \
-H "Content-Type: application/json" \
-d '{
"profile_id": "abc-123",
"text": "Hello, this is a test.",
"language": "en",
"seed": 42
}'
# Response: {"id": "gen-456", "audio_path": "/path/to/audio.wav", ...}
# Download audio
curl http://localhost:17493/audio/gen-456 -o output.wav
```
### Transcribing Audio
```bash
curl -X POST http://localhost:17493/transcribe \
-F "[email protected]" \
-F "language=en"
# Response: {"text": "Transcribed text", "duration": 5.5}
```
## Advanced Features
### Multi-Sample Profiles
Add multiple samples to a profile for better quality:
```bash
# Add first sample
curl -X POST http://localhost:17493/profiles/abc-123/samples \
-F "[email protected]" \
-F "reference_text=First sample"
# Add second sample
curl -X POST http://localhost:17493/profiles/abc-123/samples \
-F "[email protected]" \
-F "reference_text=Second sample"
# Generation will automatically combine all samples
```
### Voice Prompt Caching
Voice prompts are automatically cached for faster generation:
- First generation: ~5-10 seconds (creates prompt)
- Subsequent generations: ~1-2 seconds (uses cached prompt)
Cache is stored in `data/cache/` and persists across server restarts.
### VRAM Management
Models are lazy-loaded and can be manually unloaded:
```bash
# Unload TTS model
curl -X POST http://localhost:17493/models/unload
# Load specific model size
curl -X POST "http://localhost:17493/models/load?model_size=0.6B"
```
## Error Handling
All endpoints return proper HTTP status codes:
- `200 OK`: Success
- `400 Bad Request`: Invalid input
- `404 Not Found`: Resource not found
- `500 Internal Server Error`: Server error
Error responses include details:
```json
{
"detail": "Profile not found"
}
```
## Performance Tips
1. **Use multi-sample profiles** - Better quality than single sample
2. **Let caching work** - Voice prompts are cached automatically
3. **Use 0.6B model on CPU** - Faster than 1.7B with acceptable quality
4. **Use 1.7B model on GPU** - Best quality, still fast
5. **Unload Whisper after transcription** - Frees VRAM for TTS
## TODO
- [ ] WebSocket support for generation progress
- [ ] Batch generation endpoint
- [ ] Audio effects (M3GAN, etc.)
- [ ] Voice design (text-to-voice)
- [ ] Audio studio timeline features
- [ ] Project management
- [ ] Authentication & rate limiting
- [ ] Export/import profiles
## License
See main project LICENSE.