# Audio Transcription

Speech-to-text transcription service using faster-whisper for optimized inference.

## Features

- **High-Performance Transcription**: Uses faster-whisper (4x faster than original Whisper)
- **Multiple Model Sizes**: Support for tiny, base, small, medium, large-v2, large-v3, and large-v3-turbo
- **GPU Acceleration**: Optimized CUDA inference with int8 quantization
- **OpenAI-Compatible API**: Drop-in replacement for OpenAI's transcription endpoint
- **Multiple Output Formats**: JSON, text, and verbose JSON with segments
- **Language Detection**: Automatic language detection for 99+ languages
- **VAD Filtering**: Voice Activity Detection for improved accuracy

## Prerequisites

**Required (all modules):**
- All global prerequisites (see main [README.md](../../README.md))
- NVIDIA GPU with CUDA support (recommended for production)
- ~6GB VRAM for large-v3-turbo with int8 quantization

**Optional:**
- CPU-only mode available (slower but no GPU required)

> **Note:** The audio module runs on GPU 0 (shared with Qwen3.5-35B-A3B).

## Installation

```bash
cd modules/audio

# Install dependencies
uv sync

# Install with dev dependencies
uv sync --extra dev
```

## Quick Start

### As an API Server

```bash
cd deploy/

# Create deployment env file
cp ../.env.example .env
# Edit deploy/.env with your configuration

# Start the server
./deploy.sh --profile api --detach

# Or with nginx reverse proxy
./deploy.sh --profile api-nginx --detach
```

### API Endpoints

| Endpoint | Method | Description |
|----------|--------|-------------|
| `/health` | GET | Health check |
| `/v1/models` | GET | List available models |
| `/v1/audio/transcriptions` | POST | Transcribe audio (OpenAI-compatible) |
| `/v1/info` | GET | Service catalog metadata (used by catalog-api) |

### Example API Request

```bash
# Health check
curl http://localhost:54300/health

# Transcribe audio file
AUDIO="/path/to/audio.mp3"
curl -X POST "http://localhost:54300/v1/audio/transcriptions" \
  -F "file=@${AUDIO}" \
  -F "model=large-v3-turbo" \
  -F "response_format=json"

# With language specification and verbose output
curl -X POST "http://localhost:54300/v1/audio/transcriptions" \
  -F "file=@${AUDIO}" \
  -F "language=en" \
  -F "response_format=verbose_json"
```

### Example Response

**JSON format:**
```json
{
  "text": "This is the full transcription of your audio file.",
  "language": "en",
  "duration": 45.5
}
```

**Verbose JSON format:**
```json
{
  "text": "This is the full transcription.",
  "language": "en",
  "duration": 45.5,
  "segments": [
    {
      "id": 0,
      "start": 0.0,
      "end": 3.5,
      "text": "This is the full transcription.",
      "tokens": [123, 456, 789],
      "temperature": 0.0,
      "avg_logprob": -0.25,
      "compression_ratio": 1.5,
      "no_speech_prob": 0.01
    }
  ]
}
```

## Configuration

### Required Environment Variables

Configured via environment variables (prefix: `AUDIO_`). These are typically set in `deploy/.env`:

| Variable | Description |
|----------|-------------|
| `AUDIO_MODEL` | Whisper model name (e.g., `large-v3-turbo`) |
| `AUDIO_DEVICE` | Device: `cuda` or `cpu` |
| `AUDIO_CACHE_DIR` | Model cache directory |

### Optional Environment Variables

| Variable | Default | Description |
|----------|---------|-------------|
| `AUDIO_COMPUTE_TYPE` | `int8` | Compute type: `float16`, `int8`, `int8_float16` |
| `AUDIO_BEAM_SIZE` | `5` | Beam size for decoding (1-10) |
| `AUDIO_BEST_OF` | `5` | Number of candidates when sampling |
| `AUDIO_TEMPERATURE` | `0.0` | Sampling temperature (0.0-1.0) |
| `AUDIO_LOG_LEVEL` | `INFO` | Log level |
| `AUDIO_MAX_FILE_SIZE_MB` | `500` | Max upload file size in MB |

### Available Models

| Model | Parameters | VRAM (int8) | Speed | Quality |
|-------|-----------|-------------|-------|---------|
| `tiny` | 39M | ~1GB | 10x | Basic |
| `base` | 74M | ~1GB | 7x | Good |
| `small` | 244M | ~2GB | 4x | Better |
| `medium` | 769M | ~3-4GB | 2x | Very Good |
| `large-v3` | 1550M | ~6-8GB | 1x | Excellent |
| `large-v3-turbo` | 809M | ~6GB | 8x | Excellent |

**Recommendation:** Use `large-v3-turbo` for best balance of speed and accuracy.

## Deployment

```bash
cd deploy/

# Create deployment env file (REQUIRED)
cp ../.env.example .env
# Edit deploy/.env with your settings

# API only
./deploy.sh --profile api --detach

# API with nginx reverse proxy
./deploy.sh --profile api-nginx --detach

# View logs
./deploy.sh --profile api --logs

# Stop services
./deploy.sh --profile api --down
```

### Common Docker Commands

```bash
cd deploy/

# Restart the API service
docker compose restart audio-api

# Rebuild after code changes
docker compose build audio-api
docker compose up -d audio-api

# Check logs
docker compose logs -f audio-api
```

### Port Allocation

| Port | Service |
|------|---------|
| `54300` | Audio API (Dev + AI + Audio) — primary |
| `8200` | Audio API (legacy default) |

## Development

```bash
# Install dev dependencies
uv sync --extra dev

# Run tests
uv run pytest

# Run tests with coverage
uv run pytest --cov=src/audio --cov-report=term-missing

# Lint and format
uv run ruff check .
uv run ruff format .
```

## Architecture

```
modules/audio/
├── deploy/
│   ├── deploy.sh             # Deployment script
│   ├── docker-compose.yml    # Docker services
│   ├── Dockerfile            # Container image
│   └── nginx.conf            # Nginx reverse proxy config (optional)
├── src/audio/
│   ├── __init__.py
│   ├── app.py                # FastAPI application
│   ├── transcriber.py        # faster-whisper wrapper
│   ├── schemas.py            # Response schemas
│   └── settings.py           # Configuration
├── tests/
├── .env.example              # Environment template
├── API.md                    # API documentation
├── pyproject.toml            # Dependencies
└── README.md                 # This file
```

## Supported Audio Formats

faster-whisper (via FFmpeg) supports:
- MP3, WAV, FLAC, OGG, M4A, AAC, WMA
- MP4, AVI, MKV (audio track extraction)
- And many more formats supported by FFmpeg

## Performance Tips

1. **Use int8 quantization** for GPU inference (40% memory savings, minimal accuracy loss)
2. **Use large-v3-turbo** for best speed/accuracy tradeoff (8x faster than large-v3)
3. **Enable VAD filtering** (enabled by default) to skip silence
4. **Batch processing**: Process multiple files sequentially for better GPU utilization
5. **Language specification**: Specify language code when known for faster processing

## Troubleshooting

### Issue: Out of Memory

**Solution:** Use a smaller model or reduce compute type:
```bash
# Use medium model instead
AUDIO_MODEL=medium

# Or use float16 instead of int8 (uses more VRAM but may work better)
AUDIO_COMPUTE_TYPE=float16
```

### Issue: Slow Transcription

**Solution:**
- Ensure GPU is being used (`AUDIO_DEVICE=cuda`)
- Use int8 compute type for faster inference
- Use large-v3-turbo instead of large-v3
- Check GPU utilization with `nvidia-smi`

### Issue: Model Download Fails

**Solution:** Check network connectivity and cache directory permissions:
```bash
# Verify cache directory exists and is writable
ls -la /cai2_ds_storage/hf_cache

# Or change to local directory
AUDIO_CACHE_DIR=/home/user/.cache/huggingface
```

## References

- [faster-whisper GitHub](https://github.com/SYSTRAN/faster-whisper)
- [OpenAI Whisper](https://github.com/openai/whisper)
- [Whisper Model Card](https://github.com/openai/whisper/blob/main/model-card.md)