vocal-ai 0.3.1

Ollama-style Voice Model Management - Self-hosted OpenAI-compatible Speech AI

Vocal

Generic Speech AI Platform - Ollama for Voice Models

Vocal is an API-first speech AI platform with automatic OpenAPI spec generation, auto-generated SDK, and Ollama-style model management. Built with a generic registry pattern supporting multiple providers.

🚀 Quick Start (5 minutes)

# 1. Clone and setup
git clone <repo-url>
cd vocal
make install

# 2. Start API
make serve

# 3. Visit interactive docs
# Open: http://localhost:8000/docs

# 4. Use SDK to transcribe
python sdk_example.py your_audio.mp3

That's it! Models auto-download on first use.

Pro tip: Run make help to see all available commands.

Features

• 🎯 API-First Architecture: FastAPI with auto-generated OpenAPI spec
• 📖 Interactive Docs: Swagger UI at /docs endpoint
• 📦 Auto-Generated SDK: Python SDK generated from OpenAPI spec
• 🔄 Ollama-Style: Model registry with pull/list/delete commands
• 🚀 Fast Inference: faster-whisper (4x faster than OpenAI Whisper)
• ⚡ GPU Acceleration: Automatic CUDA detection with VRAM optimization
• 🌍 99+ Languages: Support for multilingual transcription
• 🔌 Extensible: Generic provider pattern (HuggingFace, local, custom)
• 🎤 OpenAI Compatible: /v1/audio/transcriptions endpoint
• 🔊 Text-to-Speech: Neural TTS with Piper or system voices
• 🎨 CLI Tool: Typer-based CLI with rich console output
• ✅ Production Ready: 23/23 E2E tests passing with real audio assets

Quick Start

1. Installation

git clone <repo-url>
cd vocal

# Option 1: Using Makefile (recommended)
make install

# Option 2: Using uv directly
uv venv
uv sync

2. Start API Server

# Using Makefile
make serve

# Or using uv directly
uv run uvicorn vocal_api.main:app --port 8000

# Development mode with auto-reload
make serve-dev

The API will be available at:

• API: http://localhost:8000
• Interactive Docs: http://localhost:8000/docs 🎉
• OpenAPI Spec: http://localhost:8000/openapi.json
• Health: http://localhost:8000/health

3. Use the SDK

from vocal_sdk import VocalSDK

# Initialize client
client = VocalSDK(base_url="http://localhost:8000")

# List models (Ollama-style)
models = client.models.list()
for model in models['models']:
    print(f"{model['id']}: {model['status']}")

# Download model if needed (Ollama-style pull)
client.models.download("Systran/faster-whisper-tiny")

# Transcribe audio (OpenAI-compatible)
result = client.audio.transcribe(
    file="audio.mp3",
    model="Systran/faster-whisper-tiny"
)
print(result['text'])

# Text-to-Speech
audio = client.audio.text_to_speech("Hello, world!")
with open("output.wav", "wb") as f:
    f.write(audio)

Or use the CLI:

# Transcribe audio
vocal run audio.mp3

# List models
vocal models list

# Download model
vocal models pull Systran/faster-whisper-tiny

# Start API server
vocal serve --port 8000

Or use the example:

uv run python sdk_example.py Recording.m4a

Architecture

See VOICESTACK_API_FIRST_ARCHITECTURE.md for detailed architecture documentation.

Key Principles:

• API-first design with auto-generated OpenAPI spec
• Generic registry pattern for extensibility
• Ollama-style model management
• OpenAI-compatible endpoints
• Type-safe throughout with Pydantic

vocal/
├── packages/
│   ├── core/           # Model registry & adapters ✅
│   │   └── vocal_core/
│   │       ├── registry/      # Generic model registry
│   │       │   ├── providers/ # HuggingFace, local, custom
│   │       │   └── model_info.py
│   │       └── adapters/      # STT/TTS adapters
│   │           └── stt/       # faster-whisper implementation
│   │
│   ├── api/            # FastAPI server ✅
│   │   └── vocal_api/
│   │       ├── models/        # Pydantic schemas
│   │       ├── routes/        # API endpoints
│   │       ├── services/      # Business logic
│   │       └── main.py        # FastAPI app
│   │
│   ├── sdk/            # Auto-generated Python SDK ⏳
│   └── cli/            # CLI using SDK ⏳
│
├── pyproject.toml      # uv workspace config
└── .gitignore

API Endpoints

Model Management (Ollama-style)

GET /v1/models

List all available models

Query params:

• status: Filter by status (available, downloading, not_downloaded)
• task: Filter by task (stt, tts)

GET /v1/models/{model_id}

Get model information

POST /v1/models/{model_id}/download

Download a model (Ollama-style "pull")

GET /v1/models/{model_id}/download/status

Check download progress

DELETE /v1/models/{model_id}

Delete a downloaded model

Audio Transcription (OpenAI-compatible)

POST /v1/audio/transcriptions

Transcribe audio to text.

Parameters:

• file (required): Audio file (mp3, wav, m4a, etc.)
• model (required): Model ID (e.g., "Systran/faster-whisper-tiny")
• language (optional): 2-letter language code (e.g., "en", "es")
• response_format (optional): "json" (default), "text", "srt", "vtt"
• temperature (optional): Sampling temperature (0.0-1.0, default: 0.0)

Response:

{
  "text": "Hello, how are you today?",
  "language": "en",
  "duration": 2.5,
  "segments": [
    {
      "id": 0,
      "start": 0.0,
      "end": 2.5,
      "text": "Hello, how are you today?"
    }
  ]
}

POST /v1/audio/translations

Translate audio to English text.

Text-to-Speech (OpenAI-compatible)

POST /v1/audio/speech

Convert text to speech.

Parameters:

• model (required): TTS model to use (e.g., "hexgrad/Kokoro-82M", "coqui/XTTS-v2")
• input (required): Text to synthesize
• voice (optional): Voice ID to use
• speed (optional): Speech speed multiplier (0.25-4.0, default: 1.0)
• response_format (optional): Audio format (default: "wav")

Response: Returns audio file in specified format with headers:

• X-Duration: Audio duration in seconds
• X-Sample-Rate: Audio sample rate

GET /v1/audio/voices

List available TTS voices.

Response:

{
  "voices": [
    {
      "id": "default",
      "name": "Default Voice",
      "language": "en",
      "gender": null
    }
  ],
  "total": 1
}

Health & Docs

GET /health

Health check endpoint

GET /docs

Interactive Swagger UI for API testing

GET /openapi.json

OpenAPI specification (auto-generated)

Available Models

Model ID	Size	Parameters	VRAM	Speed	Status
Systran/faster-whisper-tiny	~75MB	39M	1GB+	Fastest	CTranslate2
Systran/faster-whisper-base	~145MB	74M	1GB+	Fast	CTranslate2
Systran/faster-whisper-small	~488MB	244M	2GB+	Good	CTranslate2
Systran/faster-whisper-medium	~1.5GB	769M	5GB+	Better	CTranslate2
Systran/faster-whisper-large-v3	~3.1GB	1.5B	10GB+	Best	CTranslate2
Systran/faster-distil-whisper-large-v3	~756MB	809M	6GB+	Fast & Good	CTranslate2

All models support 99+ languages including English, Spanish, French, German, Chinese, Japanese, Arabic, and more.

Note: These use the CTranslate2-optimized models from Systran for faster-whisper, which are ~4x faster than the original OpenAI Whisper models.

Performance & Optimization

Vocal automatically detects and optimizes for your hardware:

GPU Acceleration

When NVIDIA GPU is available:

• Automatic Detection: GPU is detected and used automatically
• Optimal Compute Types:
  • 8GB+ VRAM: float16 (best quality)
  • 4-8GB VRAM: int8_float16 (balanced)
  • <4GB VRAM: int8 (most efficient)
• 4x-10x Faster: GPU inference is significantly faster than CPU
• Memory Management: Automatic GPU cache clearing

CPU Optimization

When GPU is not available:

• Multi-threading: Uses optimal CPU threads based on core count
• Quantization: int8 quantization for faster CPU inference
• VAD Filtering: Voice Activity Detection for improved performance

Check Your Device

# View device info via API
curl http://localhost:8000/v1/system/device

# Or via SDK
from vocal_sdk import VocalSDK
client = VocalSDK()
info = client._request('GET', '/v1/system/device')
print(info)

Example output:

{
  "platform": "Windows",
  "cpu_count": 16,
  "cuda_available": true,
  "gpu_count": 1,
  "gpu_devices": [{
    "name": "NVIDIA GeForce RTX 4090",
    "vram_gb": 24.0,
    "compute_capability": "8.9"
  }]
}

Optimization Tips

1. GPU Usage: Models automatically use GPU when available
2. Model Selection:
   • tiny/base models: Work well on CPU
   • small/medium: Best on GPU with 4GB+ VRAM
   • large: Requires GPU with 8GB+ VRAM
3. Batch Processing: Load model once, transcribe multiple files
4. VAD Filter: Enabled by default for better performance

CLI Usage

The CLI provides an intuitive command-line interface for common tasks.

Transcription

# Transcribe audio file
vocal run audio.mp3

# Specify model
vocal run audio.mp3 --model Systran/faster-whisper-base

# Specify language
vocal run audio.mp3 --language en

# Output formats
vocal run audio.mp3 --format text
vocal run audio.mp3 --format json
vocal run audio.mp3 --format srt
vocal run audio.mp3 --format vtt

Model Management

# List all models
vocal models list

# Filter by status
vocal models list --status available
vocal models list --status not_downloaded

# Download a model
vocal models pull Systran/faster-whisper-tiny

# Delete a model
vocal models delete Systran/faster-whisper-tiny
vocal models delete Systran/faster-whisper-tiny --force

Server Management

# Start API server (default: http://0.0.0.0:8000)
vocal serve

# Custom host and port
vocal serve --host localhost --port 9000

# Enable auto-reload for development
vocal serve --reload

Development

Project Structure

The project uses a uv workspace with multiple packages:

• packages/core: Core model registry and adapters (no dependencies on API)
• packages/api: FastAPI server (depends on core)
• packages/sdk: Auto-generated SDK (generates from API OpenAPI spec)
• packages/cli: CLI tool (uses SDK)

Running Tests

All tests use real audio assets from test_assets/audio/ with validated transcriptions.

Quick Validation (< 30 seconds)

# Using Makefile
make test-quick

# Or directly
uv run python scripts/validate.py

Full E2E Test Suite (~ 2 minutes)

# Using Makefile
make test

# With verbose output
make test-verbose

# Or using pytest directly
uv run python -m pytest tests/test_e2e.py -v

Current Status: 23/23 tests passing ✅

Test coverage includes:

• API health and device information (GPU detection)
• Model management (list, download, status, delete)
• Audio transcription with real M4A and MP3 files
• Text-to-Speech synthesis with speed control
• Error handling for invalid models and files
• Performance and model reuse optimization

Check GPU Support

make gpu-check

Code Quality

# Using Makefile
make lint          # Check code quality
make format        # Format code
make check         # Lint + format check

# Or using ruff directly
uv run ruff format .
uv run ruff check .

Makefile Commands

Vocal includes a comprehensive Makefile for common tasks:

make help          # Show all available commands

# Setup
make install       # Install dependencies
make sync          # Sync dependencies

# Testing
make test          # Run full test suite
make test-quick    # Quick validation
make test-verbose  # Verbose test output
make gpu-check     # Check GPU detection

# Development
make serve         # Start API server
make serve-dev     # Start with auto-reload
make cli           # Show CLI help
make docs          # Open API docs in browser

# Code Quality
make lint          # Run linter
make format        # Format code
make check         # Lint + format check

# Cleanup
make clean         # Remove cache files
make clean-models  # Remove downloaded models

# Quick aliases
make t             # Alias for test
make s             # Alias for serve
make l             # Alias for lint
make f             # Alias for format

Implementation Status

• ✅ Phase 0: Core Foundation

  • Generic model registry with provider pattern
  • HuggingFace provider with automatic downloads
  • faster-whisper adapter (4x faster than OpenAI)
  • Model storage & caching

• ✅ Phase 1: API Layer

  • FastAPI with auto-generated OpenAPI spec
  • Model management endpoints (Ollama-style)
  • Transcription endpoints (OpenAI-compatible)
  • Interactive Swagger UI at /docs
  • Health & status endpoints

• ✅ Phase 2: SDK

  • Auto-generated from OpenAPI spec
  • Clean Python client interface
  • Type-safe with Pydantic models
  • Namespaced APIs (models, audio)

• ✅ Phase 3: CLI

  • vocal run - Transcribe audio files
  • vocal models list/pull/delete - Model management
  • vocal serve - Start API server
  • Rich console output with progress

• ✅ Phase 4: Text-to-Speech

  • TTS API endpoints (/v1/audio/speech)
  • Multiple adapters (pyttsx3, Piper)
  • Voice selection and management
  • Speed control and audio output

• ✅ Phase 5: GPU Optimization

  • Automatic CUDA detection
  • Dynamic compute type selection (float16/int8)
  • VRAM-based optimization
  • CPU multi-threading fallback
  • System device info endpoint

• ✅ Phase 6: Testing & Production Ready

  • 23 comprehensive E2E integration tests
  • Real audio asset validation (100% accuracy)
  • Full API stack coverage
  • TTS timeout handling
  • Error handling and edge cases
  • All tests passing: 23/23 ✅

Configuration

Environment Variables

Create a .env file:

APP_NAME=Vocal API
VERSION=0.1.0
DEBUG=true
CORS_ORIGINS=["*"]
MAX_UPLOAD_SIZE=26214400

Model Storage

Models are cached at: ~/.cache/vocal/models/

Contributing

We welcome contributions! Here's how to get started:

Development Setup

# Clone the repository
git clone <repo-url>
cd vocal

# Set up environment
uv venv
uv sync

# Install packages in development mode
uv add --editable packages/core
uv add --editable packages/api
uv add --editable packages/sdk

# Run tests
uv run pytest packages/core/tests -v

Making Changes

1. Fork the repository
2. Create a feature branch: git checkout -b feature/your-feature
3. Make your changes
4. Add tests for new functionality
5. Ensure all tests pass: uv run pytest
6. Update documentation if needed
7. Commit with clear messages: git commit -m "feat: add feature X"
8. Push and create a pull request

Code Style

• Follow PEP 8 guidelines
• Use type hints for all functions
• Add docstrings for public APIs
• Keep functions focused and testable

Adding New Models

To add support for a new model provider:

1. Create a new provider class in packages/core/vocal_core/registry/providers/
2. Implement the ModelProvider interface
3. Add tests
4. Update documentation

Regenerating SDK

When API changes:

# Start API server
uv run uvicorn vocal_api.main:app --port 8000

# Download new OpenAPI spec
curl http://localhost:8000/openapi.json -o packages/sdk/openapi.json

# SDK client is hand-crafted, just update if needed

License

Server Side Public License (SSPL) v1

Vocal is open source but protects against exploitation:

• ✅ Free for personal and commercial use
• ✅ Free for self-hosting
• ✅ Free to modify and distribute
• ❌ Cannot offer as SaaS without open-sourcing your infrastructure

See LICENSE for full details.

Roadmap

• Core model registry with provider pattern
• Model management API (list, download, delete)
• SDK generation from OpenAPI spec
• Interactive Swagger UI docs
• CLI tool (Typer-based)
• Text-to-Speech (TTS) support
• Streaming transcription
• WebSocket support for real-time transcription
• Rate limiting middleware
• Authentication (optional - JWT/API keys)
• Docker deployment
• Batch transcription
• Custom model providers

Credits

Built with:

• FastAPI - Web framework
• faster-whisper - STT engine
• HuggingFace Hub - Model distribution
• uv - Python package manager