Give your AI agents the ability to listen. Microphone capture and speech-to-text.
Copy the AI prompt to install this server into Claude Code, Cursor, or another agent β or use 1-click editor setup below.
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π‘ Paste the JSON block into your client's configuration file under mcpServers, then restart the application.
Give your AI agents the ability to listen
Microphone capture and speech-to-text tools for MCP-compatible agents.
| Tool | Description |
|---|---|
list_audio_devices | List available microphone input devices |
capture_audio | Record audio from the microphone and save as WAV |
voice_query | Capture, transcribe (whisper.cpp), and query a local LLM (Ollama) |
Add to your MCP configuration:
Compatible with Claude Desktop, ChatGPT Desktop, Cursor, GitHub Copilot, Windsurf, VS Code, Gemini, Zed, and any MCP-compatible client.
Supported platforms:
Intel Mac (darwin-x64) is not supported: Apple has discontinued the platform and no decibri binary is published for it.
For list_audio_devices and capture_audio:
For voice_query (optional):
Arguments are validated before anything is recorded or written. An argument a tool does not declare is rejected with an error naming it, rather than silently ignored. duration_ms must be an integer between 100 and 30000; silence_ms an integer between 100 and 10000; stop_on_silence a boolean; device a non-negative integer index or a non-empty string id from list_audio_devices. A silence_ms that cannot take effect (passed without silence-stopping active) is rejected for the same reason unknown arguments are. A rejected call writes nothing to disk.
Returns a JSON array of available audio input devices.
Parameters: None
Example response:
The id is stable across reboots and device changes. The index is positional and can shift when devices are added or removed, and names are not unique. Prefer id when selecting a device. In the rare case the host cannot produce a stable id for a device, its id is an empty string and it can only be selected by index.
Records audio from the microphone and saves as a WAV file. Records for exactly duration_ms by default, or until the speaker stops talking with stop_on_silence: true.
Parameters:
| Parameter | Type | Default | Description |
|---|---|---|---|
duration_ms | number | 5000 | Recording duration in milliseconds (100-30000). A maximum, not an exact length, when stop_on_silence is true |
device | number or string | system default | Device index or stable device id from list_audio_devices |
stop_on_silence | boolean | false | Stop when the speaker stops talking, detected with on-device voice activity detection (Silero VAD, bundled, no download) |
silence_ms | number | 1000 | Continuous silence in milliseconds that ends a stop_on_silence recording (100-10000). Requires stop_on_silence: true |
Example response (fixed duration):
Example response (stop_on_silence: true):
With stop_on_silence, duration_ms in the response is the actual captured length, and stopped_by says how the recording ended: "silence" (the speaker finished), "ceiling" (the duration_ms maximum was reached), or "no_speech_timeout" (nobody spoke for 10 seconds; the WAV is still returned, with speech_detected: false, so silence is a reported outcome rather than an error). Detection runs per ~100ms audio buffer, so the effective hangover rounds up to the next buffer, and the recording keeps everything from the start of the call through the stop decision: nothing is gated or clipped at speech boundaries, and the audio itself is byte-identical to a fixed-duration capture of the same sounds.
Full voice pipeline: capture audio, transcribe with whisper.cpp, send to Ollama, return the response. Entirely offline. Recording stops automatically when the speaker stops talking; pass stop_on_silence: false for a fixed-length recording.
Parameters:
| Parameter | Type | Default | Description |
|---|---|---|---|
duration_ms | number | 15000 / 5000 | Maximum recording duration in milliseconds (100-30000). Default 15000 while silence-stopping is active, 5000 with stop_on_silence: false |
device | number or string | system default | Device index or stable device id from list_audio_devices |
stop_on_silence | boolean | true | Stop recording when the speaker stops talking. Pass false to record for exactly duration_ms |
silence_ms | number | 1000 | Continuous silence in milliseconds that ends the recording (100-10000) |
whisper_model | string | ggml-base.en.bin | Path or filename of Whisper GGML model |
language | string | en | Language code for transcription |
model | string | llama3.2 | Ollama model name |
prompt | string | You are a helpful assistant. | System prompt for the LLM |
Example response:
Result outcomes. voice_query reports five distinct outcomes. The structured fields are the contract (isError, speech_detected, transcription); any message is a human-readable hint whose wording is not part of the contract, so a caller branches on the fields, never on the prose. The rule is simple: if the pipeline ran, the result is a success (even when it found no words); if a dependency broke, the result is an error.
| Outcome | isError | speech_detected | transcription | response |
|---|---|---|---|---|
| Normal | absent | (true/omitted) | the text | the answer |
| No speech at all | absent | false | null | null |
| Speech, but no transcribable words | absent | true | null | null |
| Transcription step failed | true | β | β | β |
| Ollama unavailable, errored, or empty | true | β | β | β |
A caller distinguishes "the user was silent" from "the user spoke but produced no words" by speech_detected (false vs true), both carrying transcription: null. Non-speech audio never reaches the language model: whisper's non-speech markers ([BLANK_AUDIO], [MUSIC], (silence), and similar) are treated as no usable words rather than sent on as a query.
No speech (speech_detected: false):
Speech, but no transcribable words (speech_detected: true, transcription: null):
Transcription and dependency failures return isError: true with the real cause (a missing model, a whisper load failure, Ollama not running, a timeout, or an empty model response), so a caller debugging can tell whether the failure was in capture, transcription, or the language model.
mcp-listen uses decibri for cross-platform microphone capture. No ffmpeg, no SoX, no system audio tools required. Pre-built native binaries with zero setup.
Audio is captured as 16-bit PCM at 16kHz mono, the standard format for speech-to-text engines.
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