A comprehensive toolset for audio ingestion, analysis, transformation, and generation using Whisper and Librosa.
The fastest way to install a skill directly from the registry.
npx clawhub@latest install audio-processing
Copy the skill folder to one of these locations
~/.openclaw/skills/ <project>/skills/ Priority: Workspace > Local > Bundled
Copy this prompt to OpenClaw to install it automatically.
Help me install audio-processing using Clawhub. If Clawhub is not installed, install it first (npm i -g clawhub).
Get the raw skill files in a ZIP archive.
The Audio Processing skill is a versatile suite designed to empower AI agents with the ability to handle complex sound-based tasks. By integrating this capability through Openclaw Skills, developers can bridge the gap between raw audio data and actionable text or insights. It supports high-fidelity transcription via OpenAI Whisper, text-to-speech synthesis via Google TTS, and deep audio feature extraction for analytical purposes.
Security and reliability are at the forefront of this skill's design. It includes rigorous path validation to prevent unauthorized file system access and implements strict character limits for synthesis tasks. Whether you are building an automated transcription service or a voice-responsive AI assistant, these Openclaw Skills provide the necessary infrastructure to manage audio workflows efficiently and securely.
transcribe or transform by providing the required parameters like file paths or text strings.To deploy this skill, you must have FFmpeg and Python 3.8+ installed on your system. Follow these steps to install the necessary dependencies:
# Install FFmpeg (macOS example)
brew install ffmpeg
# Install required Python libraries
pip install openai-whisper gTTS librosa pydub soundfile numpy webrtcvad-wheels
The Audio Processing skill utilizes a standardized JSON schema for all inputs and outputs to ensure compatibility across Openclaw Skills workflows.
| Action | Input Parameters | Key Output Fields |
|---|---|---|
transcribe |
file_path, model |
text, segments |
tts |
text, output_path |
file_path, status |
extract_features |
file_path |
duration, sample_rate, mfcc_mean |
vad_segments |
file_path |
segments (start/end timestamps) |
transform |
file_path, ops |
file_path |
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