Figma Context MCP
Give your coding agent access to your Figma data. Implement designs in any framework in one-shot.
作者提供的说明(英文原文)
Give your AI agents the ability to listen. Microphone capture and speech-to-text.
这三项永远不会合并成一个分数:意见、测量和人气是不同的东西。
我们还没有为它写通俗的介绍。下面是我们从注册中心读取到的事实。
选择你使用的 AI 应用,然后复制设置。
你需要
你用的是哪个应用?
在上方选择你的应用。我们会记住它,下次按钮会显示“安装到 …”。
打开 Claude Desktop → Settings → Developer → Edit Config,把下面的内容粘贴到 claude_desktop_config.json
{
"mcpServers": {
"listen": {
"command": "npx",
"args": [
"-y",
"mcp-listen"
]
}
}
}在终端中运行这条命令
claude mcp add listen -- npx -y mcp-listen粘贴到项目中的 .cursor/mcp.json,或 ~/.cursor/mcp.json
{
"mcpServers": {
"listen": {
"command": "npx",
"args": [
"-y",
"mcp-listen"
]
}
}
}浏览器会先询问是否打开应用,然后应用会显示设置并请你确认。密钥的值会留空,由你自己填写。
粘贴到项目中的 .vscode/mcp.json
{
"servers": {
"listen": {
"type": "stdio",
"command": "npx",
"args": [
"-y",
"mcp-listen"
]
}
}
}浏览器会先询问是否打开应用,然后应用会显示设置并请你确认。密钥的值会留空,由你自己填写。
这个服务器在你的电脑上运行,而 ChatGPT 只能连接互联网上的服务器。请选择其他应用。
你的选择通过 Cookie 保存在这台设备上,无需账户。
这些设置是根据作者在注册中心的条目生成的。如果 README 的写法不同,请以 README 为准。
它在你的电脑上运行;安装后,工具列表会显示在你的 AI 应用中。
我们检查的是任何人都能看到的信息;我们没有阅读代码。不适用的检查项不计入总数。 · 检查于 2026年11月7日
注册名称已通过 GitHub 账户 decibri 验证,该账户拥有此仓库
我们还没有从 GitHub 读取这个仓库
我们还没有从 GitHub 读取这个仓库
它在你的电脑上运行,没有互联网端点
在你的电脑上运行,改用本地设置
它在你的电脑上运行;我们只能读取互联网端点的工具列表
与注册中心里这个分类中所有活跃的服务器比较。只有当发布者拥有该仓库时,星标数才计入。
它在 1 项适用的基础检查中通过了 1 项。我们没有阅读代码。在添加密钥或令牌之前,请先看看这个服务器要求哪些权限,并使用只具备所需权限的密钥。
注册中心没有列出必需的密钥,不过有些服务器会在首次使用时要求你登录。
它在你的电脑上运行。你的 AI 应用会在需要时启动它,数据不会经过作者的服务器(除非它自己调用在线服务)。
GitHub 仓库最后更新于–,最新版本(0.7.2)于25天前进入注册中心。
支持 MCP 的应用,例如 Claude Desktop、Claude Code、Cursor 和 VS Code。你的电脑上还需要有 Node.js。
同一分类中有:Figma Context MCP, Claude Real Video, Figma MCP Server
同一分类中其他作者的服务器,按星标数排序。
如何看卡片:★ GitHub 星标i三大支柱:评价 · 检查 · 使用情况i更新 = 最新版本发布到注册中心的时间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.
按原文从作者的仓库复制。PopNx MCP 没有审查或背书其内容。HTML、图片和脚本已被移除。链接会离开本站。
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只显示了开头部分;其余内容请在仓库中阅读