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Whissle MCP Server

A Python-based server that provides access to Whissle API endpoints for speech-to-text, diarization, translation, and text summarization.

⚠️ Important Notes

  • This server provides access to Whissle API endpoints which may incur costs

  • Each tool that makes an API call is marked with a cost warning

  • Please follow these guidelines:

    1. Only use tools when explicitly requested by the user

    2. For tools that process audio, consider the length of the audio as it affects costs

    3. Some operations like translation or summarization may have higher costs

    4. Tools without cost warnings in their description are free to use as they only read existing data

Related MCP server: audio-transcription-mcp

Prerequisites

  • Python 3.8 or higher

  • pip (Python package installer)

  • A Whissle API authentication token

Installation

  1. Clone the repository:

    git clone <repository-url>
    cd whissle_mcp
  2. Create and activate a virtual environment:

    python -m venv venv
    source venv/bin/activate  # On Windows, use: venv\Scripts\activate
  3. Install the required packages:

    pip install -e .
  4. Set up environment variables: Create a .env file in the project root with the following content:

    WHISSLE_AUTH_TOKEN=insert_auth_token_here  # Replace with your actual Whissle API token
    WHISSLE_MCP_BASE_PATH=/path/to/your/base/directory

    ⚠️ Important: Never commit your actual token to the repository. The .env file is included in .gitignore to prevent accidental commits.

  5. Configure Claude Integration: Copy claude_config.example.json to claude_config.json and update the paths:

    {
        "mcpServers": {
            "Whissle": {
                "command": "/path/to/your/venv/bin/python",
                "args": [
                    "/path/to/whissle_mcp/server.py"
                ],
                "env": {
                    "WHISSLE_AUTH_TOKEN": "insert_auth_token_here"
                }
            }
        }
    }
    • Replace /path/to/your/venv/bin/python with the actual path to your Python interpreter in the virtual environment

    • Replace /path/to/whissle_mcp/server.py with the actual path to your server.py file

Configuration

Environment Variables

  • WHISSLE_AUTH_TOKEN: Your Whissle API authentication token (required)

    • This is a sensitive credential that should never be shared or committed to version control

    • Contact your administrator to obtain a valid token

    • Store it securely in your local .env file

  • WHISSLE_MCP_BASE_PATH: Base directory for file operations (optional, defaults to user's Desktop)

Supported Audio Formats

The server supports the following audio formats:

  • WAV (.wav)

  • MP3 (.mp3)

  • OGG (.ogg)

  • FLAC (.flac)

  • M4A (.m4a)

File Size Limits

  • Maximum file size: 25 MB

  • Files larger than this limit will be rejected

Available Tools

1. Speech to Text

Convert speech to text using the Whissle API.

response = speech_to_text(
    audio_file_path="path/to/audio.wav",
    model_name="en-NER",  # Default model
    timestamps=True,      # Include word timestamps
    boosted_lm_words=["specific", "terms"],  # Words to boost in recognition
    boosted_lm_score=80   # Score for boosted words (0-100)
)

2. Speech Diarization

Convert speech to text with speaker identification.

response = diarize_speech(
    audio_file_path="path/to/audio.wav",
    model_name="en-NER",  # Default model
    max_speakers=2,       # Maximum number of speakers to identify
    boosted_lm_words=["specific", "terms"],
    boosted_lm_score=80
)

3. Text Translation

Translate text from one language to another.

response = translate_text(
    text="Hello, world!",
    source_language="en",
    target_language="es"
)

4. Text Summarization

Summarize text using an LLM model.

response = summarize_text(
    content="Long text to summarize...",
    model_name="openai",  # Default model
    instruction="Provide a brief summary"  # Optional
)

5. List ASR Models

List all available ASR models and their capabilities.

response = list_asr_models()

Response Format

Speech to Text and Diarization

{
    "transcript": "The transcribed text",
    "duration_seconds": 10.5,
    "language_code": "en",
    "timestamps": [
        {
            "word": "The",
            "startTime": 0,
            "endTime": 100,
            "confidence": 0.95
        }
    ],
    "diarize_output": [
        {
            "text": "The transcribed text",
            "speaker_id": 1,
            "start_timestamp": 0,
            "end_timestamp": 10.5
        }
    ]
}

Translation

{
    "type": "text",
    "text": "Translation:\nTranslated text here"
}

Summarization

{
    "type": "text",
    "text": "Summary:\nSummarized text here"
}

Error Response

{
    "error": "Error message here"
}

Error Handling

The server includes robust error handling with:

  • Automatic retries for HTTP 500 errors

  • Detailed error messages for different failure scenarios

  • File validation (existence, size, format)

  • Authentication checks

Common error types:

  • HTTP 500: Server error (with retry mechanism)

  • HTTP 413: File too large

  • HTTP 415: Unsupported file format

  • HTTP 401/403: Authentication error

Running the Server

  1. Start the server:

    mcp serve
  2. The server will be available at the default MCP port (usually 8000)

Testing

A test script is provided to verify the functionality of all tools:

python test_whissle.py

The test script will:

  1. Check for authentication token

  2. Test all available tools

  3. Provide detailed output of each operation

  4. Handle errors gracefully

Support

For issues or questions, please:

  1. Check the error messages for specific details

  2. Verify your authentication token

  3. Ensure your audio files meet the requirements

  4. Contact Whissle support for API-related issues

License

[Add your license information here]

Available Tools

5 tools
diarize_speechA

Convert speech to text with speaker diarization and save the output text file to a given directory. Directory is optional, if not provided, the output file will be saved to $HOME/Desktop.

⚠️ COST WARNING: This tool makes an API call to Whissle which may incur costs. Only use when explicitly requested by the user.

Args:
    audio_file_path (str): Path to the audio file to transcribe
    model_name (str, optional): The name of the ASR model to use. Defaults to "en-NER"
    max_speakers (int, optional): Maximum number of speakers to identify
    boosted_lm_words (List[str], optional): Words to boost in recognition
    boosted_lm_score (int, optional): Score for boosted words (0-100)
    output_directory (str, optional): Directory where files should be saved.
        Defaults to $HOME/Desktop if not provided.

Returns:
    TextContent with the diarized transcription and path to the output file.

Inputs: audio_file_path (required), model_name, max_speakers, boosted_lm_words, +1 more. Returns: result

list_asr_modelsB

List all available ASR models and their capabilities.

Inputs: no named parameters. Returns: Text content for a message.

speech_to_textA

Convert speech to text with a given model and save the output text file to a given directory. Directory is optional, if not provided, the output file will be saved to $HOME/Desktop.

⚠️ COST WARNING: This tool makes an API call to Whissle which may incur costs. Only use when explicitly requested by the user.

Args:
    audio_file_path (str): Path to the audio file to transcribe
    model_name (str, optional): The name of the ASR model to use. Defaults to "en-NER"
    timestamps (bool, optional): Whether to include word timestamps
    boosted_lm_words (List[str], optional): Words to boost in recognition
    boosted_lm_score (int, optional): Score for boosted words (0-100)
    output_directory (str, optional): Directory where files should be saved.
        Defaults to $HOME/Desktop if not provided.

Returns:
    TextContent with the transcription and path to the output file.

Inputs: audio_file_path (required), model_name, timestamps, boosted_lm_words, +1 more. Returns: result

summarize_textA

Summarize text using an LLM model.

⚠️ COST WARNING: This tool makes an API call to Whissle which may incur costs. Only use when explicitly requested by the user.

Args:
    content (str): The text to summarize
    model_name (str, optional): The LLM model to use. Defaults to "openai"
    instruction (str, optional): Specific instructions for summarization

Returns:
    TextContent with the summary.

Inputs: content (required), model_name, instruction. Returns: Text content for a message.

translate_textA

Translate text from one language to another.

⚠️ COST WARNING: This tool makes an API call to Whissle which may incur costs. Only use when explicitly requested by the user.

Args:
    text (str): The text to translate
    source_language (str): Source language code (e.g., "en" for English)
    target_language (str): Target language code (e.g., "es" for Spanish)

Returns:
    TextContent with the translated text.

Inputs: text (required), source_language (required), target_language (required). Returns: Text content for a message.

Tool Schema Changelog

Recent tool additions, removals, and schema changes observed during successful MCP inspections.

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    TDQS

    A3.9/5.0

    Scored across 5 tools

    Disambiguation4/5

    Most tools have distinct purposes: diarize_speech adds speaker identification to transcription, speech_to_text is basic transcription, list_asr_models provides metadata, summarize_text and translate_text handle text processing. However, diarize_speech and speech_to_text share significant overlap in core transcription functionality, which could cause confusion about when to use each.

    Naming Consistency5/5

    All tools follow a consistent verb_noun naming pattern with snake_case throughout: diarize_speech, list_asr_models, speech_to_text, summarize_text, and translate_text. The naming is predictable and follows the same grammatical structure across all five tools.

    Tool Count4/5

    Five tools is reasonable for a speech/text processing server, covering transcription (with and without diarization), model listing, summarization, and translation. The count feels slightly thin for a comprehensive MCP server but adequately covers core functionality without being overwhelming.

    Completeness3/5

    The server covers basic speech-to-text workflows and text processing operations, but has notable gaps. There's no text-to-speech capability, no audio file manipulation tools, and no batch processing operations. While the existing tools handle individual tasks, the surface feels incomplete for a comprehensive speech/text processing domain.

    Maintenance

    ActivityInactive
    ResponsivenessNo recent community issues

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