Figma Context MCP
Give your coding agent access to your Figma data. Implement designs in any framework in one-shot.
作者による説明(英語の原文)
Give your AI agents the ability to listen. Microphone capture and speech-to-text.
この3つを1つのスコアにまとめることはしません。意見、測定、人気はそれぞれ別のものだからです。
このサーバーのやさしい解説はまだ書いていません。以下はレジストリから読み取った事実です。
使っているAIアプリを選んで、設定をコピーしてください。
必要なもの
どのアプリを使っていますか?
上でアプリを選んでください。選んだアプリを覚えておくので、次からはボタンに「…にインストール」と表示されます。
Claude Desktop → Settings → Developer → Edit Config を開き、claude_desktop_config.json にこれを貼り付けます
{
"mcpServers": {
"listen": {
"command": "npx",
"args": [
"-y",
"mcp-listen"
]
}
}
}ターミナルでこれを実行します
claude mcp add listen -- npx -y mcp-listenプロジェクトの .cursor/mcp.json、または ~/.cursor/mcp.json に貼り付けます
{
"mcpServers": {
"listen": {
"command": "npx",
"args": [
"-y",
"mcp-listen"
]
}
}
}ブラウザがアプリを開いてよいか確認し、そのあとアプリが設定を表示して確認を求めます。キーの値は空欄のままなので、自分で入力してください。
プロジェクトの .vscode/mcp.json に貼り付けます
{
"servers": {
"listen": {
"type": "stdio",
"command": "npx",
"args": [
"-y",
"mcp-listen"
]
}
}
}ブラウザがアプリを開いてよいか確認し、そのあとアプリが設定を表示して確認を求めます。キーの値は空欄のままなので、自分で入力してください。
これはパソコンで動くサーバーですが、ChatGPTがつなげるのはインターネット上のサーバーだけです。ほかのアプリを選んでください。
選んだアプリは、この端末のCookieに保存されます。アカウントは不要です。
この設定は、作者のレジストリ登録情報から作成しています。READMEと違う場合は、READMEに従ってください。
パソコンで動くサーバーです。インストールすると、ツール一覧がAIアプリに表示されます。
誰でも見られる情報をチェックしており、コードは読んでいません。当てはまらない項目は数に入れていません。 · 2026年11月7日にチェック
レジストリ名は、リポジトリを所有するGitHubアカウント analyticsinmotion で確認済みです
このリポジトリはまだGitHubから読み取っていません
このリポジトリはまだGitHubから読み取っていません
これはパソコンで動くため、インターネット上のエンドポイントはありません
パソコンで動き、代わりにローカルの設定を使います
パソコンで動くサーバーです。ツール一覧を読み取れるのは、インターネット上のエンドポイントがある場合だけです
レジストリでこのカテゴリにある有効なすべてのサーバーと比較しています。スターは、公開者がリポジトリの所有者である場合だけ数えます。
当てはまる基本チェック1項目のうち1項目に合格しました。コードは読んでいません。キーやトークンを追加する前に、サーバーがどんなアクセスを求めているかを確認し、必要な権限だけを持つキーを使ってください。
レジストリには必須のキーは載っていません。ただし、最初に使うときにサインインを求めるサーバーもあります。
あなたのパソコンで動きます。必要なときにAIアプリが起動し、データは作者のサーバーを通りません(サーバー自体がオンラインサービスを呼び出す場合を除きます)。
GitHubリポジトリの最終更新は–、最新バージョン(0.1.1)がレジストリに届いたのは7 か月前です。
Claude Desktop、Claude Code、Cursor、VS CodeなどのMCP対応アプリで使えます。パソコンにNode.jsも必要です。
同じカテゴリには:Figma Context MCP, Claude Real Video, Figma MCP Server
同じカテゴリで、ほかの作者が作ったサーバー。スターの多い順。
カードの見方:★ GitHubスターi3つの柱:レビュー · チェック · 利用状況i更新=最新バージョンをレジストリに公開した日i
Give your coding agent access to your Figma data. Implement designs in any framework in one-shot.
Let any LLM watch a video locally — and search everything it has ever watched.
The Figma MCP server brings Figma design context directly into your AI workflow.
作者のリポジトリから書かれたとおりにコピーしたものです。PopNx MCPは内容を確認・推奨していません。HTML、画像、スクリプトは取り除いています。リンクはこのサイトの外へ移動します。
Give your AI agents the ability to listen
Microphone capture and speech-to-text tools for MCP-compatible agents.
Meta
| 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 on this machine (whisper.cpp), and query a model through the local Ollama daemon |
claude mcp add mcp-listen npx mcp-listen
Add to your MCP configuration:
{
"mcpServers": { "mcp-listen": { "command": "npx", "args": ["-y", "mcp-listen"] } } }
Compatible with Claude Desktop, ChatGPT Desktop, Cursor, GitHub Copilot, Windsurf, VS Code, Gemini, Zed, and any MCP-compatible client.
npm install -g mcp-listen
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:
[
{ "index": 0, "name": "Microphone", "id": "wasapi:{0.0.1.00000000}.{00000000-0000-0000-0000-000000000001}", "maxInputChannels": 2, "defaultSampleRate": 48000, "isDefault": true }, { "index": 1, "name": "Microphone Array", "id": "wasapi:{0.0.1.00000000}.{00000000-0000-0000-0000-000000000002}", "maxInputChannels": 2, "defaultSampleRate": 48000, "isDefault": false } ]
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):
{
"path": "/tmp/mcp-listen-1712345678901.wav", "duration_ms": 5000, "sample_rate": 16000, "channels": 1, "size_bytes": 160044 }
Example response (`stop_on_silence: true`):
{
"path": "/tmp/mcp-listen-1712345678901.wav", "duration_ms": 2600, "sample_rate": 16000, "channels": 1, "size_bytes": 83244, "stopped_by": "silence", "speech_detected": 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 it on this machine with whisper.cpp, send the transcription to the local Ollama daemon, and return the transcription and the response. The audio stays on this machine. 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:
{
"transcription": "What is the default port for PostgreSQL?", "response": "PostgreSQL runs on port 5432 by default.", "model": "llama3.2" }
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 | absent | the text | the answer |
| No speech at all | absent | false | null | null |
| Speech, but no transcribable words | absent | true (absent with stop_on_silence: false) | null | null |
| Transcription step failed | true | absent | absent | absent |
| Ollama unavailable, errored, or empty | true | absent | absent (the text if the reply is empty) | absent (null if the reply is empty) |
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_detected": false, "stopped_by": "no_speech_timeout", "transcription": null, "response": null, "message": "No speech was detected. Ask the user to repeat, or check that the correct microphone is selected." }
Speech, but no transcribable words (speech_detected: true, transcription: null):
{
"speech_detected": true, "stopped_by": "silence", "transcription": null, "response": null, "message": "Speech was detected but could not be transcribed. It may have been too quiet, too brief, or unclear. Ask the user to repeat, a little louder and closer to the microphone." }
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.
Silence-stopping uses the Silero voice activity detection model that ships inside decibri, running on-device through the bundled ONNX Runtime. Nothing extra is downloaded and no audio leaves the machine. The stop decision is measured in captured audio, not wall-clock time, and the VAD only decides when to stop: it never gates or alters the recorded samples.
The voice_query tool captures audio and transcribes it on this machine with whisper.cpp, sends the transcription to the local Ollama daemon, and returns the transcription and the response to the calling client. The audio stays on this machine.
The voice_query tool requires a Whisper GGML model file. Download one:
Linux / macOS:
mkdir -p ~/.mcp-listen/models
curl -L -o ~/.mcp-listen/models/ggml-base.en.bin https://huggingface.co/ggerganov/whisper.cpp/resolve/main/ggml-base.en.bin
Windows (PowerShell):
mkdir "$env:USERPROFILE\.mcp-listen\models" -Force
Invoke-WebRequest -Uri "https://huggingface.co/ggerganov/whisper.cpp/resolve/main/ggml-base.en.bin" -OutFile "$env:USERPROFILE\.mcp-listen\models\ggml-base.en.bin"
The model is ~150MB and downloads once. You can also set the WHISPER_MODEL_PATH environment variable to a custom directory.
ollama pull llama3.2ollama servesilence_ms of quiet, yielding a short capture of mostly silence. The outcome is visible, not silent: the re冒頭だけを表示しています。続きはリポジトリで読めます