Refactor documentation structure and dependencies for migration to Fumadocs

- Updated `.gitignore` to include new build and generated content directories.
- Removed outdated Mintlify configuration files and documentation.
- Introduced new `MIGRATION.md` to outline the transition from Mintlify to Fumadocs.
- Added `mdx-components.tsx` for MDX component configuration and compatibility.
- Updated `package.json` and `next.config.mjs` for new dependencies and Next.js configuration.
- Created `source.config.ts` for content source configuration.
- Added OpenAPI specification in `openapi.json` for API documentation.
- Removed legacy files and adjusted project structure to align with Fumadocs conventions.
This commit is contained in:
Jamie Pine
2026-02-02 23:29:35 -08:00
parent 788a04f265
commit 2e6efa00a2
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# Docker Deployment Guide
**Status:** In Development for v0.2.0
**Requested By:** Reddit community ([thread](https://reddit.com/r/LocalLLaMA/...))
## Overview
Docker support makes Voicebox easier to deploy, especially for:
- **Consistent Environments**: Same setup across dev/staging/prod
- **GPU Passthrough**: Easy NVIDIA/AMD GPU access
- **Server Deployments**: Run on headless Linux servers
- **Multi-User Setups**: Isolate instances per user/team
- **Cloud Platforms**: Deploy to AWS, GCP, Azure, DigitalOcean
## Quick Start
### Using Pre-Built Images (Recommended)
```bash
# CPU-only version
docker run -p 8000:8000 -v voicebox-data:/app/data \
ghcr.io/jamiepine/voicebox:latest
# NVIDIA GPU version
docker run --gpus all -p 8000:8000 -v voicebox-data:/app/data \
ghcr.io/jamiepine/voicebox:latest-cuda
# AMD GPU version (experimental)
docker run --device=/dev/kfd --device=/dev/dri -p 8000:8000 \
-v voicebox-data:/app/data \
ghcr.io/jamiepine/voicebox:latest-rocm
```
Then open: `http://localhost:8000`
### Using Docker Compose (Easiest)
Create `docker-compose.yml`:
```yaml
version: '3.8'
services:
voicebox:
image: ghcr.io/jamiepine/voicebox:latest-cuda
ports:
- "8000:8000"
volumes:
- voicebox-data:/app/data
- huggingface-cache:/root/.cache/huggingface
environment:
- GPU_MEMORY_FRACTION=0.8 # Use 80% of GPU memory
- TTS_MODE=local
- WHISPER_MODE=local
deploy:
resources:
reservations:
devices:
- driver: nvidia
count: 1
capabilities: [gpu]
volumes:
voicebox-data:
huggingface-cache:
```
Run:
```bash
docker compose up -d
```
## Building From Source
### Basic Dockerfile
```dockerfile
# Dockerfile
FROM python:3.11-slim
WORKDIR /app
# Install system dependencies
RUN apt-get update && apt-get install -y \
git \
build-essential \
ffmpeg \
&& rm -rf /var/lib/apt/lists/*
# Copy application
COPY backend/ /app/backend/
COPY requirements.txt /app/
# Install Python dependencies
RUN pip install --no-cache-dir -r requirements.txt
RUN pip install --no-cache-dir git+https://github.com/QwenLM/Qwen3-TTS.git
# Create data directory
RUN mkdir -p /app/data
# Expose port
EXPOSE 8000
# Run server
CMD ["uvicorn", "backend.main:app", "--host", "0.0.0.0", "--port", "8000"]
```
Build and run:
```bash
docker build -t voicebox .
docker run -p 8000:8000 -v $(pwd)/data:/app/data voicebox
```
### Multi-Stage Build (Optimized)
Smaller image size by separating build and runtime:
```dockerfile
# Dockerfile.optimized
# Stage 1: Build dependencies
FROM python:3.11-slim AS builder
WORKDIR /build
RUN apt-get update && apt-get install -y \
git build-essential && \
rm -rf /var/lib/apt/lists/*
COPY backend/requirements.txt .
RUN pip install --no-cache-dir --target=/build/packages \
-r requirements.txt
RUN pip install --no-cache-dir --target=/build/packages \
git+https://github.com/QwenLM/Qwen3-TTS.git
# Stage 2: Runtime
FROM python:3.11-slim
WORKDIR /app
# Install only runtime dependencies
RUN apt-get update && apt-get install -y \
ffmpeg \
&& rm -rf /var/lib/apt/lists/*
# Copy installed packages from builder
COPY --from=builder /build/packages /usr/local/lib/python3.11/site-packages/
# Copy application code
COPY backend/ /app/backend/
# Create data directory
RUN mkdir -p /app/data
EXPOSE 8000
CMD ["uvicorn", "backend.main:app", "--host", "0.0.0.0", "--port", "8000"]
```
Build:
```bash
docker build -f Dockerfile.optimized -t voicebox:slim .
```
## GPU Support
### NVIDIA GPUs (CUDA)
**Dockerfile:**
```dockerfile
FROM nvidia/cuda:12.1.0-runtime-ubuntu22.04
# Install Python
RUN apt-get update && apt-get install -y \
python3.11 python3-pip git ffmpeg && \
rm -rf /var/lib/apt/lists/*
WORKDIR /app
# Install PyTorch with CUDA support
COPY backend/requirements.txt .
RUN pip3 install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/cu121
# Install other dependencies
RUN pip3 install -r requirements.txt
RUN pip3 install git+https://github.com/QwenLM/Qwen3-TTS.git
COPY backend/ /app/backend/
EXPOSE 8000
CMD ["uvicorn", "backend.main:app", "--host", "0.0.0.0", "--port", "8000"]
```
**Run with GPU:**
```bash
docker run --gpus all -p 8000:8000 \
-v voicebox-data:/app/data \
voicebox:cuda
```
**Docker Compose with GPU:**
```yaml
services:
voicebox:
image: voicebox:cuda
deploy:
resources:
reservations:
devices:
- driver: nvidia
count: all
capabilities: [gpu]
```
### AMD GPUs (ROCm) - Experimental
**Dockerfile:**
```dockerfile
FROM rocm/dev-ubuntu-22.04:6.0
# Install Python
RUN apt-get update && apt-get install -y \
python3.11 python3-pip git ffmpeg && \
rm -rf /var/lib/apt/lists/*
WORKDIR /app
# Install PyTorch with ROCm support
COPY backend/requirements.txt .
RUN pip3 install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/rocm6.0
# Install other dependencies
RUN pip3 install -r requirements.txt
RUN pip3 install git+https://github.com/QwenLM/Qwen3-TTS.git
# Set ROCm environment variables
ENV HSA_OVERRIDE_GFX_VERSION=10.3.0
ENV ROCM_PATH=/opt/rocm
COPY backend/ /app/backend/
EXPOSE 8000
CMD ["uvicorn", "backend.main:app", "--host", "0.0.0.0", "--port", "8000"]
```
**Run with AMD GPU:**
```bash
docker run --device=/dev/kfd --device=/dev/dri \
--group-add video --ipc=host --cap-add=SYS_PTRACE \
--security-opt seccomp=unconfined \
-p 8000:8000 -v voicebox-data:/app/data \
voicebox:rocm
```
**Note:** ROCm support varies by GPU model. Works best on Linux. See [AMD ROCm docs](https://rocm.docs.amd.com) for compatibility.
## Volume Mounts
### Essential Volumes
```bash
docker run -v voicebox-data:/app/data \ # Profiles, generations, history
-v huggingface-cache:/root/.cache/huggingface \ # Downloaded models
-p 8000:8000 voicebox
```
### Development Volume Mounts
For development with hot-reload:
```bash
docker run -v $(pwd)/backend:/app/backend \ # Live code changes
-v voicebox-data:/app/data \
-e RELOAD=true \
-p 8000:8000 voicebox
```
### Custom Model Storage
Use external model directory:
```bash
docker run -v /path/to/models:/models \
-e MODELS_DIR=/models \
-v voicebox-data:/app/data \
-p 8000:8000 voicebox
```
## Environment Variables
Configure Voicebox via environment variables:
```bash
docker run -e TTS_MODE=local \
-e WHISPER_MODE=openai-api \
-e OPENAI_API_KEY=sk-... \
-e GPU_MEMORY_FRACTION=0.8 \
-e LOG_LEVEL=info \
-p 8000:8000 voicebox
```
### Available Variables
| Variable | Default | Description |
|----------|---------|-------------|
| `TTS_MODE` | `local` | TTS provider: `local`, `remote` |
| `TTS_REMOTE_URL` | - | URL for remote TTS server |
| `WHISPER_MODE` | `local` | Whisper provider: `local`, `openai-api`, `remote` |
| `WHISPER_REMOTE_URL` | - | URL for remote Whisper server |
| `OPENAI_API_KEY` | - | OpenAI API key (if using OpenAI Whisper) |
| `GPU_MEMORY_FRACTION` | `0.9` | Fraction of GPU memory to use (0.0-1.0) |
| `DATA_DIR` | `/app/data` | Directory for profiles/generations |
| `MODELS_DIR` | `/app/models` | Directory for local models |
| `LOG_LEVEL` | `info` | Logging level: `debug`, `info`, `warning`, `error` |
| `RELOAD` | `false` | Enable hot-reload for development |
## Complete Docker Compose Examples
### Production Deployment
```yaml
# docker-compose.prod.yml
version: '3.8'
services:
voicebox:
image: ghcr.io/jamiepine/voicebox:latest-cuda
container_name: voicebox
restart: unless-stopped
ports:
- "8000:8000"
volumes:
- voicebox-data:/app/data
- huggingface-cache:/root/.cache/huggingface
environment:
- TTS_MODE=local
- WHISPER_MODE=local
- GPU_MEMORY_FRACTION=0.8
- LOG_LEVEL=info
deploy:
resources:
reservations:
devices:
- driver: nvidia
count: 1
capabilities: [gpu]
healthcheck:
test: ["CMD", "curl", "-f", "http://localhost:8000/health"]
interval: 30s
timeout: 10s
retries: 3
start_period: 40s
volumes:
voicebox-data:
driver: local
huggingface-cache:
driver: local
```
Run:
```bash
docker compose -f docker-compose.prod.yml up -d
```
### Development Setup
```yaml
# docker-compose.dev.yml
version: '3.8'
services:
voicebox:
build:
context: .
dockerfile: Dockerfile
ports:
- "8000:8000"
volumes:
- ./backend:/app/backend:ro
- voicebox-data:/app/data
- huggingface-cache:/root/.cache/huggingface
environment:
- RELOAD=true
- LOG_LEVEL=debug
- TTS_MODE=local
command: uvicorn backend.main:app --host 0.0.0.0 --port 8000 --reload
volumes:
voicebox-data:
huggingface-cache:
```
### Multi-Service Stack
Full stack with reverse proxy and monitoring:
```yaml
# docker-compose.stack.yml
version: '3.8'
services:
# Main Voicebox app
voicebox:
image: ghcr.io/jamiepine/voicebox:latest-cuda
restart: unless-stopped
volumes:
- voicebox-data:/app/data
- huggingface-cache:/root/.cache/huggingface
environment:
- TTS_MODE=local
- WHISPER_MODE=local
deploy:
resources:
reservations:
devices:
- driver: nvidia
count: 1
capabilities: [gpu]
# Nginx reverse proxy
nginx:
image: nginx:alpine
ports:
- "80:80"
- "443:443"
volumes:
- ./nginx.conf:/etc/nginx/nginx.conf:ro
- ./ssl:/etc/nginx/ssl:ro
depends_on:
- voicebox
# Prometheus monitoring (optional)
prometheus:
image: prom/prometheus
ports:
- "9090:9090"
volumes:
- ./prometheus.yml:/etc/prometheus/prometheus.yml
- prometheus-data:/prometheus
volumes:
voicebox-data:
huggingface-cache:
prometheus-data:
```
## Cloud Deployment
### AWS EC2
1. **Launch GPU Instance** (g4dn.xlarge or p3.2xlarge)
2. **Install Docker + nvidia-docker:**
```bash
# Amazon Linux 2
sudo yum install -y docker
sudo systemctl start docker
distribution=$(. /etc/os-release;echo $ID$VERSION_ID)
curl -s -L https://nvidia.github.io/nvidia-docker/gpgkey | sudo apt-key add -
curl -s -L https://nvidia.github.io/nvidia-docker/$distribution/nvidia-docker.list | \
sudo tee /etc/apt/sources.list.d/nvidia-docker.list
sudo apt-get update && sudo apt-get install -y nvidia-docker2
sudo systemctl restart docker
```
3. **Deploy:**
```bash
docker run --gpus all -d -p 80:8000 \
-v voicebox-data:/app/data \
--restart unless-stopped \
ghcr.io/jamiepine/voicebox:latest-cuda
```
### DigitalOcean
Use GPU Droplet + Docker:
```bash
# Create droplet via CLI
doctl compute droplet create voicebox \
--size gpu-h100x1-80gb \
--image ubuntu-22-04-x64 \
--region nyc3
# SSH and deploy
ssh root@<droplet-ip>
curl -fsSL https://get.docker.com -o get-docker.sh
sh get-docker.sh
docker run --gpus all -d -p 80:8000 voicebox:cuda
```
### Google Cloud Run (CPU-only)
```bash
# Build and push
docker build -t gcr.io/your-project/voicebox .
docker push gcr.io/your-project/voicebox
# Deploy to Cloud Run
gcloud run deploy voicebox \
--image gcr.io/your-project/voicebox \
--platform managed \
--region us-central1 \
--memory 4Gi \
--cpu 2 \
--port 8000
```
### Fly.io
Create `fly.toml`:
```toml
app = "voicebox"
[build]
image = "ghcr.io/jamiepine/voicebox:latest"
[[services]]
http_checks = []
internal_port = 8000
protocol = "tcp"
[[services.ports]]
port = 80
handlers = ["http"]
[[services.ports]]
port = 443
handlers = ["tls", "http"]
[mounts]
source = "voicebox_data"
destination = "/app/data"
```
Deploy:
```bash
fly launch
fly deploy
```
## Troubleshooting
### GPU Not Detected
**Check NVIDIA Docker:**
```bash
docker run --rm --gpus all nvidia/cuda:12.1.0-base-ubuntu22.04 nvidia-smi
```
If this fails, reinstall nvidia-docker2.
**Check AMD ROCm:**
```bash
docker run --rm --device=/dev/kfd --device=/dev/dri rocm/dev-ubuntu-22.04:6.0 rocminfo
```
### Permission Errors
Container can't write to volumes:
```bash
# Fix permissions
docker run --user $(id -u):$(id -g) -v $(pwd)/data:/app/data voicebox
```
### Out of Memory
Reduce GPU memory usage:
```bash
docker run -e GPU_MEMORY_FRACTION=0.5 voicebox
```
Or use CPU-only:
```bash
docker run -e DEVICE=cpu voicebox
```
### Model Download Fails
Ensure HuggingFace cache is writable:
```bash
docker run -v huggingface-cache:/root/.cache/huggingface voicebox
```
Or use host cache:
```bash
docker run -v ~/.cache/huggingface:/root/.cache/huggingface voicebox
```
### Port Already in Use
Change host port:
```bash
docker run -p 8080:8000 voicebox # Use port 8080 instead
```
## Security Best Practices
### 1. Don't Run as Root
Create non-root user in Dockerfile:
```dockerfile
RUN useradd -m -u 1000 voicebox
USER voicebox
```
### 2. Use Secrets for API Keys
Don't put API keys in docker-compose.yml:
```bash
# Use Docker secrets
echo "sk-your-key" | docker secret create openai_key -
docker service create \
--secret openai_key \
-e OPENAI_API_KEY_FILE=/run/secrets/openai_key \
voicebox
```
### 3. Network Isolation
Use internal networks for multi-container setups:
```yaml
services:
voicebox:
networks:
- internal
nginx:
networks:
- internal
- external
ports:
- "80:80"
networks:
internal:
internal: true
external:
```
### 4. Resource Limits
Prevent resource exhaustion:
```yaml
services:
voicebox:
deploy:
resources:
limits:
cpus: '4'
memory: 8G
reservations:
cpus: '2'
memory: 4G
```
## Performance Tuning
### GPU Memory Management
```bash
# Use 80% of GPU (default 90%)
docker run -e GPU_MEMORY_FRACTION=0.8 voicebox
# Allow GPU memory growth (prevents OOM)
docker run -e TF_FORCE_GPU_ALLOW_GROWTH=true voicebox
```
### Model Caching
Pre-download models to volume:
```bash
# Download models first
docker run --rm -v huggingface-cache:/root/.cache/huggingface \
voicebox python -c "
from transformers import WhisperProcessor, WhisperForConditionalGeneration
WhisperProcessor.from_pretrained('openai/whisper-base')
WhisperForConditionalGeneration.from_pretrained('openai/whisper-base')
"
# Then run normally
docker run -v huggingface-cache:/root/.cache/huggingface voicebox
```
### Multi-Worker Setup
Use uvicorn workers for better throughput:
```dockerfile
CMD ["uvicorn", "backend.main:app", "--host", "0.0.0.0", "--port", "8000", "--workers", "4"]
```
## Monitoring
### Health Checks
Built-in health endpoint:
```bash
curl http://localhost:8000/health
```
Docker health check:
```yaml
healthcheck:
test: ["CMD", "curl", "-f", "http://localhost:8000/health"]
interval: 30s
timeout: 10s
retries: 3
```
### Prometheus Metrics
Add metrics exporter:
```python
# backend/main.py
from prometheus_fastapi_instrumentator import Instrumentator
Instrumentator().instrument(app).expose(app)
```
Then scrape `/metrics` with Prometheus.
### Logs
View container logs:
```bash
docker logs -f voicebox
# Or with compose
docker compose logs -f voicebox
```
## Next Steps
- [ ] Publish official images to GitHub Container Registry
- [ ] Add Kubernetes Helm charts
- [ ] Create Docker Desktop extension
- [ ] Add automated vulnerability scanning
- [ ] Support ARM64 builds for Raspberry Pi / Apple Silicon
## Contributing
Help improve Docker support:
1. Test on different platforms (AMD GPU, ARM64, etc.)
2. Submit Dockerfile optimizations
3. Share deployment configurations
4. Report issues: [GitHub Issues](https://github.com/jamiepine/voicebox/issues)
## Resources
- [Docker Documentation](https://docs.docker.com)
- [NVIDIA Container Toolkit](https://github.com/NVIDIA/nvidia-docker)
- [AMD ROCm Docker](https://rocm.docs.amd.com/projects/install-on-linux/en/latest/how-to/docker.html)
- [Docker Compose Reference](https://docs.docker.com/compose/compose-file/)
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# External Provider Support
**Status:** Planned for v0.2.0
**Discussion:** [Reddit Thread](https://reddit.com/r/LocalLLaMA/...)
## Overview
External provider support allows you to connect Voicebox to remotely-hosted TTS and Whisper services instead of running models locally. This is useful for:
- **Existing GPU Infrastructure**: You already have Qwen3-TTS running on a GPU server
- **AMD GPU Users**: Run models on your AMD hardware, use Voicebox as the UI
- **Cloud Deployments**: Host models on Modal, Replicate, RunPod, etc.
- **Team Sharing**: Multiple users share one GPU server running models
- **Mixed Deployments**: Local Whisper + remote TTS, or vice versa
## Architecture
```
┌─────────────────┐ HTTP/API ┌──────────────────┐
│ Voicebox UI │ ───────────────────────> │ Your TTS Server │
│ + Backend │ │ (Qwen3-TTS on │
│ │ <─────────────────────── │ AMD/NVIDIA GPU)│
│ - Profiles │ Audio + Metadata └──────────────────┘
│ - History │
│ - Audio Edit │ HTTP/API ┌──────────────────┐
│ - UI │ ───────────────────────> │ Whisper Service │
└─────────────────┘ │ (OpenAI API or │
│ self-hosted) │
└──────────────────┘
```
**What Voicebox Still Handles:**
- Voice profile management
- Generation history
- Audio trimming/editing
- Multi-track story editor
- UI/UX layer
**What External Providers Handle:**
- Model inference (TTS generation, transcription)
- GPU allocation
- Model loading/caching
## Configuration
### Environment Variables
```bash
# TTS Provider
TTS_MODE=remote # local | remote
TTS_REMOTE_URL=http://192.168.1.100:8000 # Your TTS server URL
TTS_API_KEY=your-api-key # Optional authentication
# Whisper Provider
WHISPER_MODE=openai-api # local | openai-api | remote
WHISPER_REMOTE_URL=http://localhost:9000 # For self-hosted Whisper
OPENAI_API_KEY=sk-... # For OpenAI Whisper API
```
### Voicebox Config UI (Planned)
Settings page will include:
- Provider selection dropdowns
- URL/API key inputs
- Connection test button
- Latency/status indicators
## Hosting External Services
### Option 1: Simple FastAPI Server (Recommended)
Create a lightweight server to expose your local Qwen3-TTS model:
```python
# tts_server.py
from fastapi import FastAPI, UploadFile, File
from qwen_tts import Qwen3TTSModel
import numpy as np
import base64
app = FastAPI()
model = Qwen3TTSModel.from_pretrained(
"Qwen/Qwen3-TTS-12Hz-1.7B-Base",
device_map="cuda" # or "cpu" for AMD ROCm: use torch+rocm
)
@app.post("/v1/generate")
async def generate(
text: str,
voice_prompt: dict,
language: str = "en",
seed: int = None
):
"""Generate speech from text using voice prompt."""
audio, sample_rate = model.generate_voice_clone(
text=text,
voice_clone_prompt=voice_prompt,
)
# Return as base64 for transport
audio_bytes = audio.tobytes()
return {
"audio": base64.b64encode(audio_bytes).decode(),
"sample_rate": sample_rate,
"dtype": str(audio.dtype)
}
@app.post("/v1/create_voice_prompt")
async def create_voice_prompt(
audio: UploadFile = File(...),
reference_text: str = ""
):
"""Create voice prompt from reference audio."""
# Save uploaded audio temporarily
audio_path = f"/tmp/{audio.filename}"
with open(audio_path, "wb") as f:
f.write(await audio.read())
# Create voice prompt
voice_prompt = model.create_voice_clone_prompt(
ref_audio=audio_path,
ref_text=reference_text,
)
return {"voice_prompt": voice_prompt}
@app.get("/health")
async def health():
return {
"status": "healthy",
"model": "Qwen3-TTS-12Hz-1.7B-Base",
"device": str(model.device)
}
if __name__ == "__main__":
import uvicorn
uvicorn.run(app, host="0.0.0.0", port=8000)
```
**Run it:**
```bash
# Install dependencies
pip install fastapi uvicorn qwen-tts torch
# For AMD GPUs, use ROCm PyTorch:
pip install torch --index-url https://download.pytorch.org/whl/rocm6.4
# Start server
python tts_server.py
```
### Option 2: vLLM (If Supported)
```bash
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-Base \
--host 0.0.0.0 \
--port 8000 \
--gpu-memory-utilization 0.9
```
### Option 3: Cloud Platforms
**Modal.com Example:**
```python
import modal
app = modal.App("qwen-tts")
image = modal.Image.debian_slim().pip_install("qwen-tts", "torch")
@app.function(gpu="A10G", image=image)
@modal.web_endpoint(method="POST")
def generate(text: str, voice_prompt: dict):
from qwen_tts import Qwen3TTSModel
model = Qwen3TTSModel.from_pretrained("Qwen/Qwen3-TTS-12Hz-1.7B-Base")
audio, sr = model.generate_voice_clone(text, voice_prompt)
return {"audio": audio.tolist(), "sample_rate": sr}
```
Deploy: `modal deploy tts_server.py`
Get URL: `https://yourapp--generate.modal.run`
## API Specification
External TTS providers must implement these endpoints:
### `POST /v1/generate`
Generate speech from text.
**Request:**
```json
{
"text": "Hello, this is a test.",
"voice_prompt": { /* voice prompt object */ },
"language": "en",
"seed": 12345
}
```
**Response:**
```json
{
"audio": "base64-encoded-audio-bytes",
"sample_rate": 24000,
"dtype": "float32"
}
```
### `POST /v1/create_voice_prompt`
Create a voice prompt from reference audio.
**Request:** (multipart/form-data)
- `audio`: Audio file upload
- `reference_text`: Transcript of the audio
**Response:**
```json
{
"voice_prompt": { /* voice prompt object */ }
}
```
### `GET /health`
Health check endpoint.
**Response:**
```json
{
"status": "healthy",
"model": "Qwen3-TTS-12Hz-1.7B-Base",
"device": "cuda:0"
}
```
## Whisper External Providers
### OpenAI Whisper API
Simply set:
```bash
WHISPER_MODE=openai-api
OPENAI_API_KEY=sk-...
```
Voicebox will use OpenAI's Whisper API automatically.
### Self-Hosted Whisper
Run your own Whisper server:
```python
# whisper_server.py
from fastapi import FastAPI, UploadFile, File
from transformers import WhisperProcessor, WhisperForConditionalGeneration
import librosa
app = FastAPI()
processor = WhisperProcessor.from_pretrained("openai/whisper-base")
model = WhisperForConditionalGeneration.from_pretrained("openai/whisper-base")
@app.post("/v1/transcribe")
async def transcribe(audio: UploadFile = File(...), language: str = None):
# Load audio
audio_path = f"/tmp/{audio.filename}"
with open(audio_path, "wb") as f:
f.write(await audio.read())
audio_data, sr = librosa.load(audio_path, sr=16000)
# Process
inputs = processor(audio_data, sampling_rate=16000, return_tensors="pt")
predicted_ids = model.generate(inputs["input_features"])
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
return {"text": transcription}
```
Configure Voicebox:
```bash
WHISPER_MODE=remote
WHISPER_REMOTE_URL=http://localhost:9000
```
## Use Cases
### 1. AMD GPU User with Existing Setup
**Scenario:** You have a Radeon 7900 XTX running Qwen3-TTS on Linux.
**Setup:**
1. Run `tts_server.py` on your AMD box (ROCm PyTorch)
2. Configure Voicebox: `TTS_MODE=remote`, `TTS_REMOTE_URL=http://amd-box:8000`
3. Use Voicebox UI for profiles, generation, editing
4. TTS happens on your AMD GPU
### 2. Team Deployment
**Scenario:** 5 team members, 1 GPU server.
**Setup:**
1. Deploy TTS server on shared GPU box
2. Each person runs Voicebox desktop app locally
3. All point to same `TTS_REMOTE_URL`
4. Profiles and history stay local per user
5. GPU usage is shared
### 3. Hybrid Local/Remote
**Scenario:** Fast local Whisper, heavy TTS on cloud.
**Setup:**
```bash
TTS_MODE=remote
TTS_REMOTE_URL=https://your-modal-app.modal.run
WHISPER_MODE=local # Fast transcription on your CPU
```
### 4. OpenAI Whisper + Self-Hosted TTS
**Scenario:** Use OpenAI's API for transcription, run TTS locally.
**Setup:**
```bash
TTS_MODE=local
WHISPER_MODE=openai-api
OPENAI_API_KEY=sk-...
```
## Security Considerations
### Authentication
Add API key authentication to your external server:
```python
from fastapi import Header, HTTPException
API_KEY = "your-secret-key"
async def verify_api_key(x_api_key: str = Header(...)):
if x_api_key != API_KEY:
raise HTTPException(status_code=401, detail="Invalid API key")
@app.post("/v1/generate", dependencies=[Depends(verify_api_key)])
async def generate(...):
...
```
Configure Voicebox:
```bash
TTS_API_KEY=your-secret-key
```
### Network Security
- **VPN/Tailscale**: Use private network for remote servers
- **HTTPS**: Use reverse proxy (nginx/Caddy) with SSL certificates
- **Firewall**: Restrict access to known IPs
### Rate Limiting
Protect your external server:
```python
from slowapi import Limiter
from slowapi.util import get_remote_address
limiter = Limiter(key_func=get_remote_address)
app.state.limiter = limiter
@app.post("/v1/generate")
@limiter.limit("10/minute")
async def generate(...):
...
```
## Performance Considerations
### Latency
External providers add network latency:
- **Local network**: ~10-50ms overhead (negligible)
- **Same datacenter**: ~1-5ms overhead
- **Cross-region cloud**: 50-200ms+ overhead
For real-time applications, keep TTS server on local network or same cloud region.
### Caching
Implement response caching on external server:
```python
from functools import lru_cache
@lru_cache(maxsize=1000)
def get_cached_generation(text, voice_prompt_hash, language, seed):
return model.generate_voice_clone(text, voice_prompt)
```
### Load Balancing
For high-traffic deployments, run multiple TTS servers behind a load balancer:
```
Voicebox ──> Load Balancer ──> TTS Server 1 (GPU 1)
├──> TTS Server 2 (GPU 2)
└──> TTS Server 3 (GPU 3)
```
## Future Enhancements
- [ ] **Provider Marketplace**: Built-in directory of compatible providers
- [ ] **Automatic Fallback**: If remote fails, fallback to local
- [ ] **Cost Tracking**: Monitor API usage and costs
- [ ] **Performance Metrics**: Latency, throughput dashboards
- [ ] **Multi-Provider**: Use different providers for different voices/languages
## Contributing
If you build an external provider, please share:
1. Server implementation
2. Performance benchmarks
3. Deployment guide
Submit to: [GitHub Discussions](https://github.com/jamiepine/voicebox/discussions)
## Questions?
- **Discord**: [Join the community](https://discord.gg/...)
- **GitHub**: [Open an issue](https://github.com/jamiepine/voicebox/issues)
- **Docs**: [Full documentation](https://voicebox.sh/docs)
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# MLX Audio Integration
**Status:** Validated ✅
**Context:** [mlx-audio v0.3.1 release](https://github.com/Blaizzy/mlx-audio)
## Validation Results
We validated mlx-audio in an isolated environment (`mlx-test/`). Key findings:
| Metric | Result |
|--------|--------|
| MLX Version | 0.30.4 |
| Model Load Time | ~1s (after initial download) |
| Generation RTF | **0.5-0.6x** (1.7-2x faster than real-time) |
| Test Hardware | Apple Silicon Mac |
### Model Mapping
| voicebox (PyTorch) | mlx-audio (MLX) |
|--------------------|-----------------|
| `Qwen/Qwen3-TTS-12Hz-1.7B-Base` | `mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16` |
| `Qwen/Qwen3-TTS-12Hz-0.6B-Base` | (not yet converted) |
### mlx-audio API
The API uses a **generator-based streaming pattern**:
```python
from mlx_audio.tts import load
model = load("mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16")
# generate() yields GenerationResult objects
for result in model.generate("Hello world"):
audio = result.audio # numpy array of samples
sample_rate = result.sample_rate # 24000
rtf = result.real_time_factor # e.g., 0.55
```
### Known Warnings (harmless)
```
You are using a model of type qwen3_tts to instantiate a model of type .
The tokenizer you are loading... with an incorrect regex pattern...
```
These warnings appear but don't affect functionality or output quality.
### Demo Script
Run `mlx-test/demo.py` to test:
```bash
cd mlx-test && source venv/bin/activate && python demo.py "Your text here"
```
## Problem
Apple Silicon users are stuck on CPU inference while Windows and Linux users get CUDA acceleration. The current PyTorch MPS backend has stability issues (lines 34-36 in `backend/tts.py` and `backend/transcribe.py`), forcing a CPU fallback that makes voicebox significantly slower on M1/M2/M3 Macs.
This creates a poor experience for a large portion of users who bought Apple Silicon specifically for ML workloads.
## Solution
Integrate [mlx-audio](https://github.com/Blaizzy/mlx-audio) as the inference engine for macOS Apple Silicon builds. MLX is Apple's native ML framework, optimized for Metal and the unified memory architecture. It's fast, stable, and already supports the same Qwen3-TTS models we use.
**Key wins:**
- Native GPU acceleration on Apple Silicon (no more CPU fallback)
- Streaming TTS support (faster perceived latency)
- Memory optimizations (run larger models on less RAM)
- Fixed 0.6B silence bug that we currently ship
- Same Qwen3-TTS models (zero migration cost for users)
## Architecture
### Current Stack
```
┌─────────────────────────┐
│ PyTorch + Qwen3-TTS │
│ (CPU only on macOS) │
└─────────────────────────┘
```
### Proposed Stack
```
┌─────────────────────────────────────────┐
│ Platform Detection at Runtime │
└─────────────────────────────────────────┘
├─── Apple Silicon (aarch64-darwin)
│ ┌─────────────────────────┐
│ │ MLX Audio Backend │
│ │ - Qwen3-TTS (mlx) │
│ │ - Whisper (mlx) │
│ │ - Streaming support │
│ └─────────────────────────┘
└─── Other (x86_64, Windows, Linux)
┌─────────────────────────┐
│ PyTorch Backend │
│ - Qwen3-TTS (pytorch) │
│ - Whisper (pytorch) │
│ - CUDA if available │
└─────────────────────────┘
```
## Implementation Phases
### Phase 1: Platform Detection & Dependency Management
Create a backend that switches between PyTorch and MLX based on runtime platform detection.
**New files:**
- `backend/platform.py` - Detect Apple Silicon, return backend type
- `backend/backends/__init__.py` - Backend factory pattern
- `backend/requirements-mlx.txt` - MLX-specific deps (macOS only)
**Modified files:**
- `backend/requirements.txt` - Keep PyTorch as default
- `backend/main.py` - Import from backend factory instead of direct imports
**Platform detection logic:**
```python
def get_backend_type() -> str:
"""Detect best backend for current platform."""
if platform.system() == "Darwin" and platform.machine() == "arm64":
# Apple Silicon detected
try:
import mlx
return "mlx"
except ImportError:
return "pytorch" # Fallback if mlx not installed
return "pytorch"
```
### Phase 2: MLX Backend Implementation
Create parallel implementations of TTS and STT using mlx-audio.
**New files:**
- `backend/backends/mlx_backend.py` - MLX inference engine
- `backend/backends/pytorch_backend.py` - Refactor current code into backend
**Interface both backends must implement:**
```python
class TTSBackend(Protocol):
async def load_model(self, model_size: str) -> None: ...
async def create_voice_prompt(self, audio_path: str, reference_text: str) -> dict: ...
async def generate(self, text: str, voice_prompt: dict, **kwargs) -> Tuple[np.ndarray, int]: ...
async def generate_streaming(self, text: str, voice_prompt: dict, **kwargs) -> AsyncIterator[bytes]: ...
def unload_model(self) -> None: ...
class STTBackend(Protocol):
async def load_model(self, model_size: str) -> None: ...
async def transcribe(self, audio_path: str, language: Optional[str]) -> str: ...
def unload_model(self) -> None: ...
```
**MLX backend implementation notes:**
mlx-audio's `generate()` returns a generator by default (streaming is built-in):
```python
# MLX backend wrapper
from mlx_audio.tts import load
class MLXTTSBackend:
def __init__(self):
self.model = None
async def load_model(self, model_size: str) -> None:
model_map = {
"1.7B": "mlx-community/Qwen3-TTS-12Hz-1.7B-Base-bf16",
# "0.6B": needs conversion to mlx format
}
self.model = load(model_map[model_size])
async def generate(self, text: str, voice_prompt: dict, **kwargs) -> Tuple[np.ndarray, int]:
# Collect all chunks from generator
chunks = []
for result in self.model.generate(text): # TODO: add voice_prompt support
chunks.append(np.array(result.audio))
return np.concatenate(chunks), 24000
```
**MLX-specific features to expose:**
- Streaming TTS (new endpoint: `/api/generate/stream`)
- Memory-optimized model loading
- Qwen3-ASR for transcription (in addition to Whisper)
### Phase 3: API Layer Updates
Update FastAPI endpoints to support new streaming capabilities and maintain backward compatibility.
**Modified files:**
- `backend/main.py` - Add streaming endpoints
- `backend/tts.py` - Refactor to use backend abstraction
- `backend/transcribe.py` - Refactor to use backend abstraction
**New endpoints:**
```python
@app.post("/api/generate/stream")
async def generate_stream(...) -> StreamingResponse:
"""Stream TTS chunks as they're generated (MLX only)."""
backend = get_backend()
if not hasattr(backend, 'generate_streaming'):
raise HTTPException(501, "Streaming not supported on this backend")
return StreamingResponse(backend.generate_streaming(...), media_type="audio/wav")
```
**Backward compatibility:**
- Keep all existing `/api/generate` endpoints unchanged
- PyTorch backend users see no behavior change
- MLX users automatically get faster inference, streaming is opt-in
### Phase 4: Frontend Integration
Add UI indicators for backend type and streaming progress.
**Modified files:**
- `app/src/hooks/useGenerationForm.tsx` - Add streaming support
- `app/src/components/GenerationForm.tsx` - Show backend badge, streaming toggle
- `app/src/lib/api.ts` - Add streaming API client
**UI additions:**
- Badge showing current backend ("MLX" or "PyTorch")
- Toggle for streaming mode (disabled if PyTorch)
- Real-time streaming playback (WaveSurfer progressive loading)
### Phase 5: Build & Distribution
Create separate installers for MLX (Apple Silicon) and PyTorch (Universal).
**Modified files:**
- `tauri/src-tauri/tauri.conf.json` - Add target-specific builds
- `.github/workflows/release.yml` - Build both variants
**Build matrix:**
```yaml
- target: aarch64-apple-darwin
backend: mlx
installer: voicebox-macos-silicon-{version}.dmg
- target: x86_64-apple-darwin
backend: pytorch
installer: voicebox-macos-intel-{version}.dmg
- target: x86_64-pc-windows-msvc
backend: pytorch
installer: voicebox-windows-{version}.exe
```
**Installation flow:**
- Auto-detect architecture, recommend correct installer
- MLX installer includes `mlx-audio` in embedded Python
- PyTorch installer includes `torch` in embedded Python
- Both can coexist (different backend, same profile format)
### Phase 6: Testing & Validation
Ensure both backends produce compatible outputs.
**New files:**
- `backend/tests/test_backend_parity.py` - Verify both backends produce similar audio
- `backend/tests/test_streaming.py` - Streaming-specific tests
**Test scenarios:**
- Same voice prompt on both backends → similar (not identical) audio output
- Profile created on MLX → loads on PyTorch (and vice versa)
- Streaming chunks assemble into valid WAV file
- Model downloads work on both backends
- Memory usage stays within bounds
### Phase 7: Documentation
Update user-facing docs and developer guides.
**New files:**
- `docs/developer/BACKENDS.md` - Guide for adding new backends
- `docs/overview/performance.md` - Backend comparison benchmarks
**Modified files:**
- `README.md` - Note Apple Silicon acceleration
- `docs/TROUBLESHOOTING.md` - Add MLX-specific issues
**Key docs to write:**
- Which installer to download (architecture detection)
- Performance comparison (MLX vs PyTorch on same M2 hardware)
- How streaming mode works
- How to force PyTorch on Apple Silicon (for debugging)
## Technical Decisions
### Why Dual Backend Instead of MLX-Only?
**Pros of dual backend:**
- Windows and Intel Mac users unaffected
- Easier testing (can compare outputs)
- Fallback if MLX has issues
**Cons of dual backend:**
- More code to maintain
- Two dependency trees
- Build complexity (separate installers)
**Decision:** Dual backend. The maintenance cost is worth it to avoid breaking existing users and to have a fallback.
### Why Separate Installers Instead of Runtime Detection?
**Pros of separate installers:**
- Smaller bundle size (don't ship both PyTorch and MLX)
- Clearer to users which version they have
- Easier to debug (no "which backend am I running?" confusion)
- Can optimize each build for its target
**Cons:**
- More installers to build and test
- Users might download the wrong one
**Decision:** Separate installers. Bundle size matters (PyTorch + MLX would be huge), and we can auto-detect architecture on the download page.
### Streaming vs Batch Generation
MLX supports streaming, PyTorch doesn't (without significant work). Should streaming be:
1. MLX-only feature (✅ chosen)
2. Implemented for both (lots of work)
3. Not exposed at all (wasted opportunity)
**Decision:** MLX-only. Expose as opt-in feature with graceful degradation (button disabled on PyTorch backend).
## Migration Path
Nothing needs migrating, macos users will just notice a speed-boost in inference
**Data format compatibility:**
- Profiles (SQLite) → no schema changes needed
- Voice prompts (cached) → backend-agnostic (just numpy arrays)
- Audio files → unchanged
## Performance Expectations
### Measured Results (from validation)
| Metric | MLX (measured) | PyTorch CPU (estimated) |
|--------|----------------|-------------------------|
| **6s audio generation** | ~3-4s | ~10-15s |
| **Real-time factor** | 0.5-0.6x | 2-3x |
| **Model load (cached)** | ~1s | ~3-5s |
### TTS Generation (1.7B model, ~20s output)
- **PyTorch CPU (M2 Max):** ~45-60s (slower than real-time)
- **MLX (M2 Max):** ~8-12s (faster than real-time)
- **Improvement:** ~4-5x faster
### Whisper Transcription (10s audio clip)
- **PyTorch CPU:** ~5-8s
- **MLX:** ~1-2s
- **Improvement:** ~3-4x faster
### Memory Usage (1.7B model)
- **PyTorch:** ~8-10GB (no GPU offload, so CPU RAM)
- **MLX:** ~4-6GB (unified memory, better optimization)
- **Improvement:** ~40% less RAM
Full benchmarks will be in `docs/overview/performance.md` after Phase 6.
## Open Questions
- **Should we support Qwen3-ASR (MLX-only) in addition to Whisper?** Adds another model option but increases complexity. Probably phase 8+. - Sure
- **Should we backport streaming to PyTorch?** Would require chunking and callback-based generation. Probably not worth it given mlx-audio already has it. - No
- **What's the auto-update UX for migrating PyTorch→MLX users?** Needs design. Don't want to force reinstall, but also want to make upgrade obvious. - it just updates, users see nothing
- **Do we expose backend selection in settings or hide it?** Leaning toward auto-detect only, with env var override for power users.
## Success Metrics
How we'll know this worked:
1. **Performance:** Apple Silicon users report generation faster than real-time
2. **Adoption:** >80% of macOS downloads are MLX build within 1 month
3. **Stability:** <5% increase in bug reports (backend abstraction doesn't introduce regressions)
4. **Feedback:** Positive sentiment in Discord/GitHub about macOS performance
## Related Work
- [PyTorch MPS tracking issue](https://github.com/pytorch/pytorch/issues/77764) - Why we can't use MPS directly
- [mlx-audio server implementation](https://github.com/Blaizzy/mlx-audio/blob/main/examples/server.py) - Reference for streaming API
- [MLX Whisper benchmarks](https://github.com/ml-explore/mlx-examples/tree/main/whisper) - Performance data
## Next Steps
1. ~~Validate mlx-audio can load Qwen3-TTS models (quick test)~~ ✅ Done - see `mlx-test/`
2. Get approval on dual-backend architecture
3. Start Phase 1 (platform detection)
## Questions?
Feedback welcome in GitHub discussions or Discord.
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# OpenAI API Compatibility
**Status:** Planned for v0.2.0
**Issue:** [#10 OpenAI API compatibility](https://github.com/jamiepine/voicebox/issues/10)
## Overview
This feature exposes OpenAI-compatible endpoints from Voicebox, allowing any tool, library, or application that speaks the OpenAI Audio API to use Voicebox as a drop-in local replacement.
```mermaid
flowchart LR
subgraph clients [External Clients]
SDK[OpenAI SDK]
Curl[curl / HTTP]
Apps[Third-party Apps]
end
subgraph voicebox [Voicebox Server]
OpenAI["/v1/audio/* endpoints"]
TTS[TTSModel]
Whisper[WhisperModel]
Profiles[Voice Profiles]
end
SDK --> OpenAI
Curl --> OpenAI
Apps --> OpenAI
OpenAI --> TTS
OpenAI --> Whisper
OpenAI --> Profiles
```
## Use Cases
- **OpenAI SDK users**: `openai.audio.speech.create()` works with Voicebox
- **LLM frameworks**: LangChain, AutoGen, etc. can use Voicebox for TTS
- **Shell scripts**: `curl` commands copy-pasted from OpenAI docs work
- **Existing integrations**: Any tool expecting OpenAI's API works without code changes
## Endpoints to Implement
### 1. `POST /v1/audio/speech` (TTS)
OpenAI spec: https://platform.openai.com/docs/api-reference/audio/createSpeech
**Request:**
```json
{
"model": "tts-1",
"input": "Hello world!",
"voice": "alloy",
"response_format": "mp3",
"speed": 1.0
}
```
**Response:** Audio file (mp3, wav, opus, aac, flac, pcm)
**Voice Mapping Strategy:**
- `voice` parameter maps to Voicebox profile names (case-insensitive)
- If no match, use a configurable default profile
- Support special syntax: `voice: "profile:uuid"` for explicit profile ID
### 2. `POST /v1/audio/transcriptions` (Whisper)
OpenAI spec: https://platform.openai.com/docs/api-reference/audio/createTranscription
**Request:** (multipart/form-data)
- `file`: Audio file
- `model`: "whisper-1"
- `language`: Optional language hint
- `response_format`: json, text, srt, verbose_json, vtt
**Response:**
```json
{
"text": "Hello world!"
}
```
## Implementation Details
### New File: `backend/openai_compat.py`
Create a dedicated module with an APIRouter for OpenAI-compatible endpoints:
```python
from fastapi import APIRouter, UploadFile, File, Form, HTTPException
from fastapi.responses import StreamingResponse
from pydantic import BaseModel
from typing import Literal, Optional
router = APIRouter(prefix="/v1/audio", tags=["OpenAI Compatible"])
class SpeechRequest(BaseModel):
model: str = "tts-1"
input: str
voice: str = "alloy"
response_format: Literal["mp3", "wav", "opus", "aac", "flac", "pcm"] = "mp3"
speed: float = 1.0
@router.post("/speech")
async def create_speech(request: SpeechRequest, db: Session = Depends(get_db)):
# 1. Map voice name to profile
# 2. Generate audio using existing TTSModel
# 3. Convert to requested format
# 4. Return audio stream
...
@router.post("/transcriptions")
async def create_transcription(
file: UploadFile = File(...),
model: str = Form("whisper-1"),
language: Optional[str] = Form(None),
response_format: str = Form("json"),
):
# 1. Save uploaded file
# 2. Transcribe using existing WhisperModel
# 3. Return in requested format
...
```
### Voice Profile Resolution
Add helper in [backend/profiles.py](backend/profiles.py):
```python
async def resolve_voice_for_openai(voice: str, db: Session) -> Optional[VoiceProfile]:
"""
Resolve OpenAI voice parameter to a Voicebox profile.
Priority:
1. Exact profile name match (case-insensitive)
2. Profile ID match (if voice starts with "profile:")
3. Default profile from config
4. First available profile
"""
...
```
### Audio Format Conversion
Add conversion utilities in [backend/utils/audio.py](backend/utils/audio.py):
```python
def convert_audio_format(
audio: np.ndarray,
sample_rate: int,
target_format: str, # mp3, wav, opus, aac, flac, pcm
) -> bytes:
"""Convert audio to target format using ffmpeg or pydub."""
...
```
### Configuration
Add to [backend/config.py](backend/config.py):
```python
# OpenAI API Compatibility
OPENAI_COMPAT_ENABLED = True
OPENAI_COMPAT_DEFAULT_VOICE = None # Profile ID or name for default voice
OPENAI_COMPAT_REQUIRE_AUTH = False # Require API key validation
OPENAI_COMPAT_API_KEY = None # If set, validate against this
```
### Integration with main.py
In [backend/main.py](backend/main.py), include the router:
```python
from . import openai_compat
# Add OpenAI-compatible routes
if config.OPENAI_COMPAT_ENABLED:
app.include_router(openai_compat.router)
```
## Streaming Support (Future Enhancement)
Initial implementation returns complete audio. Streaming can be added later:
```python
@router.post("/speech")
async def create_speech(request: SpeechRequest):
if request.stream:
return StreamingResponse(
generate_audio_chunks(request),
media_type=f"audio/{request.response_format}"
)
...
```
## Testing
Example usage after implementation:
```bash
# TTS with curl
curl http://localhost:8000/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{"model": "tts-1", "input": "Hello!", "voice": "MyProfile"}' \
--output speech.mp3
# With OpenAI Python SDK
from openai import OpenAI
client = OpenAI(base_url="http://localhost:8000/v1", api_key="unused")
response = client.audio.speech.create(
model="tts-1",
voice="MyProfile",
input="Hello world!"
)
response.stream_to_file("output.mp3")
# Transcription
curl http://localhost:8000/v1/audio/transcriptions \
-F file=@audio.mp3 \
-F model="whisper-1"
```
## Security Considerations
- Optional API key validation (for shared deployments)
- Rate limiting on endpoints
- Input length limits (same as existing `/generate` endpoint)
## Dependencies
- `pydub` or `ffmpeg-python` for audio format conversion (mp3, opus, etc.)
- No changes to existing TTS/Whisper model code
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# TTS Provider Architecture
**Status:** Planned for v0.1.13
**Created:** 2025-01-31
**Problem:** GitHub 2GB release limit + poor UX for frequent updates requiring 2.4GB re-downloads
---
## Overview
Split the monolithic backend into modular components:
1. **Main App** (~150-200MB): Tauri + FastAPI backend + Whisper + UI/profiles/history
2. **TTS Providers** (downloadable plugins): Separate executables for model inference
This architecture solves:
- ✅ GitHub 2GB release artifact limit
- ✅ Frequent app updates without re-downloading large python binaries
- ✅ User choice of compute backend (CPU/GPU/Cloud)
- ✅ External provider support (OpenAI, custom servers)
- ✅ Future extensibility
---
## Architecture Diagram
```
┌─────────────────────────────────────────────────────────┐
│ Voicebox App (Tauri + Backend) ~150MB │
│ ├─ UI Layer (React) │
│ ├─ Backend (FastAPI) │
│ │ ├─ Voice Profiles │
│ │ ├─ Generation History │
│ │ ├─ Audio Editing / Stories │
│ │ └─ Provider Manager ◄──────────────┐ │
│ └─ Whisper (bundled, tiny ~50MB) │ │
└─────────────────────────────────────────┼────────────────┘
HTTP/IPC │
┌────────────────────────────────┼─────────────────┐
│ │ │
▼ ▼ ▼
┌─────────────────┐ ┌─────────────────┐ ┌──────────────────┐
│ TTS Provider: │ │ TTS Provider: │ │ TTS Provider: │
│ PyTorch CPU │ │ PyTorch CUDA │ │ MLX (Apple) │
│ │ │ │ │ │
│ ~300MB │ │ ~2.4GB │ │ ~800MB │
│ │ │ │ │ │
│ Local inference │ │ GPU inference │ │ Metal inference │
└─────────────────┘ └─────────────────┘ └──────────────────┘
│ │ │
└────────────────────────┴─────────────────────┘
┌─────────────▼──────────────┐
│ Future Providers: │
│ • Remote Server │
│ • OpenAI API │
│ • ElevenLabs │
│ • Custom Docker Container │
└────────────────────────────┘
```
---
## Problem Statement
### Current Architecture Issues
**Monolithic Binary:**
- CPU version: ~295MB
- CUDA version: ~2.37GB
- GitHub releases: 2GB file size limit (BLOCKED)
- Updates require re-downloading entire binary
- Poor UX: update app → restart → download CUDA update → restart again
**User Pain Points:**
1. Cannot release CUDA version on GitHub (over 2GB)
2. Every app update forces 2.4GB re-download for GPU users
3. No flexibility (can't use OpenAI, remote servers, etc.)
4. Wastes bandwidth for small bug fixes
---
## Solution: Pluggable TTS Providers
### Component Breakdown
#### 1. Main App (voicebox.exe / .app / .AppImage)
**Size:** ~100-150MB
**Includes:**
- Tauri runtime + React UI
- FastAPI backend (pure Python, no PyTorch)
- Whisper model (tiny, ~50MB)
- SQLite database
- Profile/history/audio editing logic
- Provider management system
**Does NOT include:**
- PyTorch (CPU or CUDA)
- TTS models (Qwen3-TTS)
- Heavy ML dependencies
**Updates frequently:** UI fixes, feature additions, non-ML changes
---
#### 2. TTS Provider: PyTorch CPU
**Binary:** `tts-provider-pytorch-cpu.exe`
**Size:** ~200MB
**Includes:**
- PyTorch CPU build
- Qwen3-TTS package
- Transformers
- No CUDA libraries
**Download source:** Cloudflare R2
**Updates rarely:** Only when model code changes
---
#### 3. TTS Provider: PyTorch CUDA
**Binary:** `tts-provider-pytorch-cuda.exe`
**Size:** ~2.4GB
**Includes:**
- PyTorch CUDA build (cu121)
- Qwen3-TTS package
- CUDA runtime, cuDNN, cuBLAS
- Transformers
**Download source:** Cloudflare R2
**Platform:** Windows + Linux (NVIDIA GPU)
**Updates rarely:** Only when model code or CUDA version changes
---
#### 4. TTS Provider: MLX
**Binary:** `tts-provider-mlx`
**Size:** ~150MB
**Includes:**
- MLX framework
- MLX-optimized Qwen3-TTS
- Metal acceleration
**Platform:** macOS only (Apple Silicon)
**Download source:** Cloudflare R2
---
#### 5. TTS Provider: Remote
**Binary:** None (built-in config)
**Size:** 0MB
**How it works:**
- User provides URL to their own TTS server
- Backend proxies requests to that server
- Implements API spec from `EXTERNAL_PROVIDERS.md`
**Use cases:**
- AMD GPU users running their own server
- Team deployments with shared GPU server
- Cloud hosting (Modal, RunPod, Replicate)
---
#### 6. TTS Provider: OpenAI
**Binary:** None (API wrapper)
**Size:** 0MB
**How it works:**
- User provides OpenAI API key
- Backend wraps OpenAI Audio API
- Voice profiles map to OpenAI voices
**Benefits:**
- Zero local compute
- Pay-per-use
- Instant setup
---
## Communication Protocol
### Provider API Specification
All TTS providers must implement these endpoints:
#### POST /tts/generate
Generate speech from text.
**Request:**
```json
{
"text": "Hello world!",
"voice_prompt": {
/* voice prompt object */
},
"language": "en",
"seed": 12345,
"model_size": "1.7B"
}
```
**Response:**
```json
{
"audio": "base64-encoded-audio",
"sample_rate": 24000,
"duration": 2.5
}
```
#### POST /tts/create_voice_prompt
Create voice prompt from reference audio.
**Request:** (multipart/form-data)
- `audio`: Audio file
- `reference_text`: Transcript
**Response:**
```json
{
"voice_prompt": {
/* serialized prompt */
}
}
```
#### GET /tts/health
Health check.
**Response:**
```json
{
"status": "healthy",
"provider": "pytorch-cuda",
"version": "1.0.0",
"model": "Qwen3-TTS-12Hz-1.7B-Base",
"device": "cuda:0"
}
```
#### GET /tts/status
Model status.
**Response:**
```json
{
"model_loaded": true,
"model_size": "1.7B",
"available_sizes": ["0.6B", "1.7B"],
"gpu_available": true,
"vram_used_mb": 1234
}
```
---
## Backend Implementation
### Provider Manager
**File:** `backend/providers/__init__.py`
```python
class ProviderManager:
"""Manages TTS provider lifecycle."""
def __init__(self):
self.active_provider: Optional[Provider] = None
self.config = load_provider_config()
async def start_provider(self, provider_type: str) -> str:
"""Start a TTS provider process."""
if provider_type == "pytorch-cpu":
return await self._start_local_provider("tts-provider-pytorch-cpu.exe")
elif provider_type == "pytorch-cuda":
return await self._start_local_provider("tts-provider-pytorch-cuda.exe")
elif provider_type == "mlx":
return await self._start_local_provider("tts-provider-mlx")
elif provider_type == "remote":
return self.config["remote_url"]
elif provider_type == "openai":
return None # No subprocess, API wrapper
async def _start_local_provider(self, binary_name: str) -> str:
"""Start local provider subprocess."""
provider_path = get_provider_binary_path(binary_name)
if not provider_path.exists():
raise ProviderNotInstalledException(binary_name)
# Start subprocess on random port
port = get_free_port()
process = subprocess.Popen([
str(provider_path),
"--port", str(port),
"--data-dir", str(config.get_data_dir())
])
# Wait for provider to be ready
await wait_for_provider_health(f"http://localhost:{port}")
self.active_provider = Provider(process, port)
return f"http://localhost:{port}"
async def stop_provider(self):
"""Stop active provider."""
if self.active_provider:
self.active_provider.process.terminate()
self.active_provider = None
```
---
### Provider Abstraction
**File:** `backend/providers/base.py`
```python
class TTSProvider(ABC):
"""Abstract base for TTS providers."""
@abstractmethod
async def generate(
self,
text: str,
voice_prompt: dict,
language: str,
seed: Optional[int]
) -> tuple[np.ndarray, int]:
"""Generate speech audio."""
pass
@abstractmethod
async def create_voice_prompt(
self,
audio_path: str,
reference_text: str
) -> dict:
"""Create voice prompt from reference audio."""
pass
```
**File:** `backend/providers/local.py`
```python
class LocalProvider(TTSProvider):
"""Provider that communicates with local subprocess via HTTP."""
def __init__(self, base_url: str):
self.base_url = base_url
self.client = httpx.AsyncClient()
async def generate(self, text, voice_prompt, language, seed):
response = await self.client.post(
f"{self.base_url}/tts/generate",
json={
"text": text,
"voice_prompt": voice_prompt,
"language": language,
"seed": seed
}
)
data = response.json()
audio = np.frombuffer(base64.b64decode(data["audio"]), dtype=np.float32)
return audio, data["sample_rate"]
```
**File:** `backend/providers/openai.py`
```python
class OpenAIProvider(TTSProvider):
"""Provider that wraps OpenAI Audio API."""
def __init__(self, api_key: str):
self.client = OpenAI(api_key=api_key)
async def generate(self, text, voice_prompt, language, seed):
# Map voice_prompt to OpenAI voice name
voice = map_profile_to_openai_voice(voice_prompt)
response = await self.client.audio.speech.create(
model="tts-1",
voice=voice,
input=text
)
# Convert to numpy array
audio_data = response.content
audio, sr = load_audio_from_bytes(audio_data)
return audio, sr
```
---
## Provider Installation
### Download Manager
**File:** `backend/providers/installer.py`
```python
class ProviderInstaller:
"""Handles provider download and installation."""
async def download_provider(self, provider_type: str):
"""Download provider binary from R2."""
binary_name = {
"pytorch-cpu": "tts-provider-pytorch-cpu.exe",
"pytorch-cuda": "tts-provider-pytorch-cuda.exe",
"mlx": "tts-provider-mlx"
}[provider_type]
download_url = f"https://downloads.voicebox.sh/providers/v{PROVIDER_VERSION}/{binary_name}"
# Download with progress tracking (reuse existing SSE system)
await download_with_progress(
url=download_url,
destination=get_provider_install_path(binary_name),
progress_key=f"provider-{provider_type}"
)
```
**Provider Storage Location:**
- Windows: `%APPDATA%/voicebox/providers/`
- macOS: `~/Library/Application Support/voicebox/providers/`
- Linux: `~/.local/share/voicebox/providers/`
---
## Frontend Implementation
### Provider Settings UI
**Component:** `app/src/components/ServerSettings/ProviderSettings.tsx`
```tsx
export function ProviderSettings() {
const [selectedProvider, setSelectedProvider] =
useState<ProviderType>("auto");
const {data: installedProviders} = useQuery({
queryKey: ["providers", "installed"],
queryFn: () => apiClient.getInstalledProviders(),
});
return (
<Card>
<CardHeader>
<CardTitle>TTS Provider</CardTitle>
<CardDescription>Choose how Voicebox generates speech</CardDescription>
</CardHeader>
<CardContent>
<RadioGroup
value={selectedProvider}
onValueChange={setSelectedProvider}
>
{/* Auto-detect */}
<div className="flex items-center space-x-2">
<RadioGroupItem value="auto" id="auto" />
<Label htmlFor="auto">
<div className="font-medium">Auto-detect (Recommended)</div>
<div className="text-sm text-muted-foreground">
Automatically choose the best available provider
</div>
</Label>
</div>
{/* PyTorch CUDA */}
<div className="flex items-center justify-between">
<div className="flex items-center space-x-2">
<RadioGroupItem
value="pytorch-cuda"
id="cuda"
disabled={!gpuAvailable}
/>
<Label htmlFor="cuda">
<div className="font-medium">PyTorch CUDA (NVIDIA GPU)</div>
<div className="text-sm text-muted-foreground">
4-5x faster inference on NVIDIA GPUs
</div>
</Label>
</div>
{!installedProviders?.includes("pytorch-cuda") && gpuAvailable && (
<Button
onClick={() => downloadProvider("pytorch-cuda")}
size="sm"
>
Download (2.4GB)
</Button>
)}
</div>
{/* PyTorch CPU */}
<div className="flex items-center justify-between">
<div className="flex items-center space-x-2">
<RadioGroupItem value="pytorch-cpu" id="cpu" />
<Label htmlFor="cpu">
<div className="font-medium">PyTorch CPU</div>
<div className="text-sm text-muted-foreground">
Works on any system, slower inference
</div>
</Label>
</div>
{!installedProviders?.includes("pytorch-cpu") && (
<Button onClick={() => downloadProvider("pytorch-cpu")} size="sm">
Download (300MB)
</Button>
)}
</div>
{/* MLX (macOS only) */}
{isMacOS && (
<div className="flex items-center justify-between">
<div className="flex items-center space-x-2">
<RadioGroupItem value="mlx" id="mlx" />
<Label htmlFor="mlx">
<div className="font-medium">MLX (Apple Silicon)</div>
<div className="text-sm text-muted-foreground">
Optimized for M1/M2/M3 chips
</div>
</Label>
</div>
{!installedProviders?.includes("mlx") && (
<Button onClick={() => downloadProvider("mlx")} size="sm">
Download (800MB)
</Button>
)}
</div>
)}
{/* Remote */}
<div className="space-y-2">
<div className="flex items-center space-x-2">
<RadioGroupItem value="remote" id="remote" />
<Label htmlFor="remote">
<div className="font-medium">Remote Server</div>
<div className="text-sm text-muted-foreground">
Connect to your own TTS server
</div>
</Label>
</div>
{selectedProvider === "remote" && (
<Input placeholder="http://your-server:8000" className="ml-6" />
)}
</div>
{/* OpenAI */}
<div className="space-y-2">
<div className="flex items-center space-x-2">
<RadioGroupItem value="openai" id="openai" />
<Label htmlFor="openai">
<div className="font-medium">OpenAI API</div>
<div className="text-sm text-muted-foreground">
Use OpenAI's TTS API (requires API key)
</div>
</Label>
</div>
{selectedProvider === "openai" && (
<Input type="password" placeholder="sk-..." className="ml-6" />
)}
</div>
</RadioGroup>
</CardContent>
</Card>
);
}
```
---
## File Structure
```
voicebox/
├── backend/
│ ├── main.py # Main FastAPI app (no TTS code)
│ ├── providers/
│ │ ├── __init__.py # ProviderManager
│ │ ├── base.py # TTSProvider ABC
│ │ ├── local.py # LocalProvider (subprocess)
│ │ ├── remote.py # RemoteProvider (HTTP)
│ │ ├── openai.py # OpenAIProvider (API wrapper)
│ │ └── installer.py # Provider download logic
│ ├── profiles.py # Voice profile management
│ ├── history.py # Generation history
│ ├── transcribe.py # Whisper (still bundled)
│ └── ... (other backend modules)
├── providers/
│ ├── pytorch-cpu/
│ │ ├── main.py # FastAPI server for TTS
│ │ ├── tts_backend.py # PyTorch TTS logic
│ │ ├── requirements.txt # torch (CPU), qwen-tts, transformers
│ │ └── build.spec # PyInstaller spec
│ │
│ ├── pytorch-cuda/
│ │ ├── main.py # FastAPI server for TTS
│ │ ├── tts_backend.py # PyTorch TTS logic
│ │ ├── requirements.txt # torch+cu121, qwen-tts, transformers
│ │ └── build.spec # PyInstaller spec
│ │
│ └── mlx/
│ ├── main.py # FastAPI server for TTS
│ ├── mlx_backend.py # MLX TTS logic
│ ├── requirements.txt # mlx, qwen-tts-mlx
│ └── build.spec # PyInstaller spec
├── app/ # Frontend (Tauri + React)
│ └── src/
│ └── components/
│ └── ServerSettings/
│ └── ProviderSettings.tsx
└── tauri/
└── src-tauri/
└── tauri.conf.json # No externalBin for providers
```
---
## Migration Path
### Phase 1: Refactor Backend (No User Changes)
**Goal:** Abstract TTS behind provider interface
1. Create `backend/providers/` module structure
2. Implement `TTSProvider` abstract base class
3. Create `LocalProvider` wrapper for current PyTorch code
4. Modify `backend/tts.py` to use provider abstraction
5. Keep PyTorch bundled in main app
**Result:** Code is prepared, but user experience unchanged
---
### Phase 2: Build Provider Binaries
**Goal:** Create standalone TTS provider executables
1. Create separate PyInstaller specs for each provider
2. Build provider executables:
- `tts-provider-pytorch-cpu.exe` (~300MB)
- `tts-provider-pytorch-cuda.exe` (~2.4GB)
- `tts-provider-mlx` (~800MB, macOS)
3. Test subprocess communication
4. Upload providers to Cloudflare R2
**Result:** Provider binaries exist but aren't used yet
---
### Phase 3: Remove PyTorch from Main App
**Goal:** Split main app from providers
1. Exclude PyTorch/Qwen3-TTS from main app PyInstaller spec
2. Main app now requires provider download
3. Update GitHub CI to build multiple artifacts:
- `voicebox-{version}-{platform}.exe` (~150MB)
- `tts-provider-pytorch-cpu-{version}.exe`
- `tts-provider-pytorch-cuda-{version}.exe`
- `tts-provider-mlx-{version}` (macOS)
**Result:** Main app is small, providers downloaded separately
---
### Phase 4: Add Provider UI
**Goal:** User-facing provider management
1. Create Provider Settings page
2. Implement provider download UI
3. Add provider status indicators
4. Show active provider in UI
**Result:** Users can choose and download providers
---
### Phase 5: External Providers
**Goal:** Enable remote and cloud providers
1. Implement `RemoteProvider` (HTTP client)
2. Implement `OpenAIProvider` (API wrapper)
3. Add provider configuration UI (URLs, API keys)
4. Document external provider API spec
**Result:** Full provider ecosystem
---
## Provider Versioning
### Independent Versioning
Providers have their own version numbers, independent of the main app:
- **App version:** `v0.2.0` (frequent updates)
- **Provider version:** `v1.0.0` (rare updates)
### Compatibility Matrix
**Example:**
| App Version | Min Provider Version | Max Provider Version |
| ----------- | -------------------- | -------------------- |
| v0.2.0 | v1.0.0 | v1.x.x |
| v0.3.0 | v1.0.0 | v1.x.x |
| v0.4.0 | v1.2.0 | v1.x.x |
| v1.0.0 | v2.0.0 | v2.x.x |
**Backend checks compatibility:**
```python
async def check_provider_compatibility(provider_version: str) -> bool:
"""Check if provider version is compatible with current app."""
min_version = "1.0.0"
max_version = "1.999.999"
return min_version <= provider_version < max_version
```
**UI shows warning if incompatible:**
```
⚠️ Provider version 0.9.0 is outdated. Update to v1.0.0+
```
---
## User Flows
### First-Time Setup
1. User downloads and installs Voicebox (~150MB)
2. App launches → detects no TTS provider installed
3. Shows setup wizard:
```
Choose your TTS provider:
[ ] PyTorch CUDA (2.4GB) [Download]
✓ Fastest on NVIDIA GPUs
✗ Requires NVIDIA GPU
[●] PyTorch CPU (300MB) [Download]
✓ Works on any system
✗ Slower inference
[ ] MLX (800MB) [Download]
✓ Fast on Apple Silicon
✗ macOS only (M1/M2/M3)
[ ] Remote Server
URL: ___________________
[ ] OpenAI API
API Key: ________________
```
4. User selects provider → downloads with progress bar
5. Provider installs to AppData/Application Support
6. App starts provider → ready to use
---
### App Update Flow (No Provider Change)
**Scenario:** Bug fix in UI, no backend changes
1. User gets update notification: "Voicebox v0.2.1 available"
2. Downloads update (~150MB, not 2.4GB!)
3. Installs and restarts
4. **Provider stays the same** (no re-download needed)
5. App starts using existing provider
**User experience:** Fast updates, no multi-GB downloads
---
### Provider Update Flow
**Scenario:** New Qwen3-TTS model version released
1. User opens Settings → Provider tab
2. Sees notification: "Provider update available (v1.1.0)"
3. Clicks "Update Provider"
4. Downloads new provider binary
5. Old provider binary is replaced
6. Restart app to use new provider
**Frequency:** Rare (only when TTS model/backend changes)
---
### Switching Providers
**Scenario:** User upgrades to NVIDIA GPU
1. User goes to Settings → Provider
2. Selects "PyTorch CUDA"
3. Clicks "Download" → downloads 2.4GB
4. Download completes → restarts app
5. App now uses CUDA provider
---
## Benefits
| Benefit | Details |
| ----------------------------- | --------------------------------------------------------- |
| **GitHub Releases Work** | Main app ~150MB << 2GB limit |
| **Fast Updates** | UI/feature updates don't require re-downloading providers |
| **User Choice** | CPU, CUDA, MLX, OpenAI, remote server |
| **External Provider Support** | Users can run their own TTS servers |
| **Bandwidth Savings** | Only download provider once, app updates are small |
| **Future-Proof** | Easy to add new providers (ElevenLabs, custom models) |
| **Team Deployments** | Multiple users share one remote provider |
| **Cloud-Ready** | Works with Modal, Replicate, RunPod, etc. |
---
## Open Questions
### 1. Provider Versioning
**Question:** Should providers have independent versions or match app version?
**Options:**
- A. Independent (providers: v1.x, app: v0.2.x)
- B. Matched (both use v0.2.x)
**Recommendation:** Independent versioning with compatibility matrix
---
### 2. Auto-Update Providers
**Question:** Should providers auto-update separately from app?
**Options:**
- A. Manual updates only (user clicks "Update Provider")
- B. Optional auto-update (user can enable)
- C. Always auto-update
**Recommendation:** Optional auto-update (default off)
---
### 3. Provider Discovery
**Question:** How does app find installed providers?
**Options:**
- A. Check standard paths in AppData/Application Support
- B. Registry (Windows) / plist (macOS)
- C. Config file with provider locations
**Recommendation:** Standard paths + config fallback
---
### 4. Fallback Behavior
**Question:** What if no provider is installed?
**Options:**
- A. Show setup wizard on first launch
- B. Block app until provider installed
- C. Allow app to run in "demo mode" (transcription only)
**Recommendation:** Setup wizard on first launch
---
### 5. Provider Auto-Start
**Question:** Should provider start automatically with app?
**Options:**
- A. Always start selected provider on app launch
- B. Start on-demand (when user generates speech)
- C. User preference
**Recommendation:** Auto-start (configurable in settings)
---
## Future Enhancements
- [ ] **Provider Marketplace:** Built-in directory of community providers
- [ ] **Multi-Provider Support:** Use different providers per voice/language
- [ ] **Provider Health Monitoring:** Automatic failover if provider crashes
- [ ] **Cost Tracking:** Monitor API usage for OpenAI/cloud providers
- [ ] **Performance Metrics:** Latency, throughput, VRAM usage dashboards
- [ ] **Docker Providers:** Run providers in Docker containers
- [ ] **Provider Plugins:** Load custom providers from user scripts
---
## Related Documents
- [EXTERNAL_PROVIDERS.md](./EXTERNAL_PROVIDERS.md) - External provider support plan
- [OPENAI_SUPPORT.md](./OPENAI_SUPPORT.md) - OpenAI API compatibility
- [github-2gb-limit-issue.md](../github-2gb-limit-issue.md) - Original problem
- [r2-setup.md](../r2-setup.md) - Cloudflare R2 configuration
---
## Contributing
If you want to build a custom TTS provider:
1. Implement the provider API spec (see above)
2. Test with Voicebox locally
3. Package as executable (PyInstaller, Docker, etc.)
4. Share in GitHub Discussions
**Questions?**
- GitHub Issues: [voicebox/issues](https://github.com/jamiepine/voicebox/issues)
- Discord: Coming soon