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# Whisper API server
A local, OpenAI-compatible speech recognition API service using the Whisper model. This service provides a straightforward way to transcribe audio files in various formats with high accuracy and is designed to be compatible with the OpenAI Whisper API.
![Client Interface](client.png)
## Features
- 🔊 High-quality speech recognition using Whisper models
- 🌐 OpenAI-compatible API endpoints
- 🚀 Hardware acceleration support (CUDA, MPS)
- ⚡ Flash Attention 2 for faster transcription on compatible GPUs
- 🎛️ Audio preprocessing for better transcription results
- 🔄 Multiple input methods (file upload, URL, base64, local files)
- 📊 Optional timestamp generation for word-level alignment
- 🎧 Convenient built-in client with text editing and audio playback capabilities
- 📱 Responsive web interface included
- 📝 Transcription history logging
## Requirements
- Python 3.12+ recommended
- CUDA-compatible GPU (optional, for faster processing)
- FFmpeg and SoX for audio processing
- Whisper model (download from Hugging Face)
## Installation
### Using server.sh (recommended)
1. Clone the repository:
```bash
git clone https://github.com/kreolsky/whisper-api-server.git
cd whisper-api-server
```
2. Run the server script with the update flag to create and set up the conda environment:
```bash
chmod +x server.sh
./server.sh --update
```
This will:
- Create a conda environment named "whisper-api" with Python 3.12
- Install all required dependencies
- Start the service
### Manual installation
1. Create and activate a conda environment:
```bash
conda create -n whisper-api python=3.12
conda activate whisper-api
```
2. Install the required dependencies:
```bash
pip install -r requirements.txt
```
3. Start the service:
```bash
python server.py
```
## Docker
The repository now includes a container setup for running the API in Docker.
### What gets mounted
- `./config.docker.json` -> `/app/config.docker.json`
- `./logs` -> `/app/logs`
- `./history` -> `/app/history`
- `./models` -> `/models`
Place your Whisper model inside `./models/whisper` or update `model_path` in `config.docker.json`.
That path must contain the actual Hugging Face model files, not just an empty folder.
If you prefer automatic bootstrap, set:
```json
"auto_download_missing_model": true,
"model_repo_id": "openai/whisper-large-v3"
```
With the default Docker config, the container will automatically download that
model into `/models/whisper` on first start if the directory is missing or incomplete.
Expected example layout:
```text
models/
whisper/
config.json
generation_config.json
preprocessor_config.json
tokenizer.json
tokenizer_config.json
model.safetensors
```
### Build and run with Docker Compose
If your selected model requires authentication on Hugging Face, export a token first:
```bash
export HF_TOKEN=your_huggingface_token
```
```bash
docker compose up --build
```
The API will be available at:
```text
http://localhost:5042
```
### Build and run with plain Docker
```bash
docker build -t whisper-api-server .
docker run --rm -p 5042:5042 \
-e HF_TOKEN="$HF_TOKEN" \
-v "$(pwd)/config.docker.json:/app/config.docker.json:ro" \
-v "$(pwd)/logs:/app/logs" \
-v "$(pwd)/history:/app/history" \
-v "$(pwd)/models:/models" \
whisper-api-server
```
### GPU note
The container image installs the same Python dependencies as the local setup, including CUDA-oriented PyTorch wheels on Linux x86_64. To actually use NVIDIA GPU acceleration at runtime, start the container with GPU access enabled in your Docker environment, for example:
```bash
docker run --rm --gpus all -p 5042:5042 \
-e HF_TOKEN="$HF_TOKEN" \
-v "$(pwd)/config.docker.json:/app/config.docker.json:ro" \
-v "$(pwd)/logs:/app/logs" \
-v "$(pwd)/history:/app/history" \
-v "$(pwd)/models:/models" \
whisper-api-server
```
## Configuration
The service is configured through the `config.json` file:
```json
{
"service_port": 5042,
"model_path": "/path/to/whisper/model",
"auto_download_missing_model": false,
"model_repo_id": "openai/whisper-large-v3",
"language": "russian",
"enable_history": true,
"max_history_days": 30,
"chunk_length_s": 28,
"batch_size": 6,
"max_new_tokens": 384,
"temperature": 0.01,
"return_timestamps": false,
"audio_rate": 16000,
"norm_level": "-0.55",
"compand_params": "0.3,1 -90,-90,-70,-50,-40,-15,0,0 -7 0 0.15",
"device_id": 0,
"file_validation": {
"max_file_size_mb": 500,
"allowed_extensions": [".wav", ".mp3", ".ogg", ".flac", ".m4a", ".oga", ".aac", ".webm"],
"allowed_mime_types": ["audio/wav", "audio/mpeg", "audio/ogg", "audio/flac", "audio/mp4", "audio/x-m4a", "audio/aac", "audio/webm"]
},
"log_level": "INFO",
"log_file": "logs/whisper_api.log",
"request_logging": {
"exclude_endpoints": ["/health", "/static"]
}
}
```
### Configuration parameters
| Parameter | Description |
|-----------|-------------|
| `service_port` | Port on which the service will run |
| `model_path` | Path to the Whisper model directory |
| `auto_download_missing_model` | Download the configured fallback model into `model_path` when the local directory is missing or incomplete |
| `model_repo_id` | Hugging Face model repo to download when auto-download is enabled |
| `language` | Language for transcription (e.g., "russian", "english") |
| `enable_history` | Whether to save transcription history (true/false) |
| `max_history_days` | Number of days to keep transcription history before rotation |
| `chunk_length_s` | Length of audio chunks for processing (in seconds) |
| `batch_size` | Batch size for processing |
| `max_new_tokens` | Maximum new tokens for the model output |
| `temperature` | Model temperature parameter (lower = more deterministic) |
| `return_timestamps` | Whether to return timestamps in the transcription |
| `audio_rate` | Audio sampling rate in Hz |
| `norm_level` | Normalization level for audio preprocessing |
| `compand_params` | Parameters for audio compression/expansion |
| `device_id` | CUDA device index to use for inference |
| `file_validation.max_file_size_mb` | Maximum allowed file size in megabytes |
| `file_validation.allowed_extensions` | List of accepted audio file extensions |
| `file_validation.allowed_mime_types` | List of accepted MIME types |
| `log_level` | Logging level (DEBUG, INFO, WARNING, ERROR) |
| `log_file` | Path to the log file |
| `request_logging.exclude_endpoints` | Endpoints excluded from request logging |
## Web interface
The service includes a user-friendly web interface accessible at:
```
http://localhost:5042/
```
The interface allows you to:
- Upload audio files via drag-and-drop or file picker
- Upload multiple files for sequential processing
- Listen to the uploaded audio
- Edit the transcription text if needed
- Download results as TXT or JSON or copy results to clipboard
- View API request/response details for debugging
## API usage
### Health check
```bash
curl http://localhost:5042/health
```
### Get configuration
```bash
curl http://localhost:5042/config
```
### Get available models
```bash
curl http://localhost:5042/v1/models
```
### Transcribe an audio file (OpenAI-compatible)
```bash
curl -X POST http://localhost:5042/v1/audio/transcriptions \
-F file=@audio.mp3
```
### Transcribe from URL
```bash
curl -X POST http://localhost:5042/v1/audio/transcriptions/url \
-H "Content-Type: application/json" \
-d '{"url":"https://example.com/audio.mp3"}'
```
### Transcribe from base64
```bash
curl -X POST http://localhost:5042/v1/audio/transcriptions/base64 \
-H "Content-Type: application/json" \
-d '{"file":"base64_encoded_audio_data"}'
```
### Transcribe asynchronously
Submit a file for background transcription and receive a task ID:
```bash
curl -X POST http://localhost:5042/v1/audio/transcriptions/async \
-F file=@audio.mp3
```
Response:
```json
{"task_id": "abc123..."}
```
### Get async task status
```bash
curl http://localhost:5042/v1/tasks/<task_id>
```
Response when completed:
```json
{"task_id": "abc123...", "status": "completed", "result": {...}}
```
Possible statuses: `pending`, `completed`, `failed`.
### Request with additional parameters
```bash
curl -X POST http://localhost:5042/v1/audio/transcriptions \
-F file=@audio.mp3 \
-F language=english \
-F return_timestamps=true \
-F temperature=0.0
```
## Response format
### Without timestamps
```json
{
"text": "Transcribed text content",
"processing_time": 2.34,
"response_size_bytes": 1234,
"duration_seconds": 10.5,
"model": "whisper-large-v3"
}
```
### With timestamps
```json
{
"segments": [
{
"start_time_ms": 0,
"end_time_ms": 5000,
"text": "First segment of text"
},
{
"start_time_ms": 5000,
"end_time_ms": 10000,
"text": "Second segment of text"
}
],
"text": "First segment of text Second segment of text",
"processing_time": 3.45,
"response_size_bytes": 2345,
"duration_seconds": 10.5,
"model": "whisper-large-v3"
}
```
## Advanced usage
### Using with different models
You can use any Whisper model by changing the `model_path` in the configuration:
1. Download a model from Hugging Face
2. Update the `model_path` in `config.json`
3. Restart the service
The recommended model for Russian speech recognition is [whisper-large-v3-russian-ties-podlodka-v1.2](https://huggingface.co/Apel-sin/whisper-large-v3-russian-ties-podlodka-v1.2).
### Hardware acceleration
The service automatically selects the best available compute device:
- CUDA GPU (device index configured via `device_id` in `config.json`)
- Apple Silicon MPS (for Mac with M1/M2/M3 chips)
- CPU (fallback)
For best performance on NVIDIA GPUs, Flash Attention 2 is used when available.
### Transcription history
When `enable_history` is set to `true`, transcription results are saved in a `history` folder organized by date. Each transcription is saved as a JSON file with the format:
```
history/
└── YYYY-MM-DD/
└── timestamp_filename_xxxx.json
```
## Troubleshooting
### Audio processing issues
If you encounter audio processing errors:
- Ensure that FFmpeg and SoX are installed on your system
- Check that the audio file is not corrupted
- Try different audio preprocessing parameters in the configuration
### Performance issues
For slow transcription:
- Use a GPU if available
- Adjust `chunk_length_s` and `batch_size` parameters
- Consider using a smaller Whisper model
- Reduce `audio_rate` if full quality isn't needed