list_audio_filesLABSCO SUMMARY
Finding the file is half the work, and this does that half.
Most transcription tools want a path. If you record regularly you do not have a path, you have a vague memory of a long conversation some time last week — which is exactly what the filtering is for. The enhancement templates are worth knowing about before you settle for a raw transcript: the same recording turned into an analytical summary or a professional write-up is usually closer to what you wanted. Run convert_audio first if you plan to ask questions about the audio, since chat only accepts mp3 and wav.
What it is
An MCP server over a local directory of audio files, backed by OpenAI's transcription, audio-chat and speech models. The part that makes it usable is the file layer: you can find the recording by describing it — duration, modified time, size — instead of knowing its filename.
What you get
- list_audio_files — filter and sort the directory by duration, modified time or size, with each file annotated by which models can handle it
- get_latest_audio — the most recent recording, for when you just finished making one
- convert_audio to mp3 or wav, and compress_audio for files over the size limit — which is what stops a long recording from being rejected
- transcribe_audio — the plain transcript
- transcribe_with_enhancement — the same audio through a template: detailed keeps tone, emotion and background; storytelling turns it into narrative; professional produces business-appropriate text; analytical adds analysis of speech patterns and key points
- chat_with_audio — ask questions about a recording rather than reading its transcript, using the audio models directly
- create_audio — text to speech, with a voice, an optional instruction on how to deliver it, and a speed
- Transcription accepts flac, mp3, mp4, mpeg, mpga, m4a, ogg, wav and webm; audio chat is limited to mp3 and wav, which is what convert_audio is for
Requirements
An OpenAI API key in OPENAI_API_KEY, and AUDIO_FILES_PATH pointing at the directory you want addressable. Install with uv from the repository. You pay OpenAI per minute transcribed and per character spoken; the server itself adds nothing. Your audio is uploaded to OpenAI when a tool runs — the file management happens locally, the model work does not.
Setup effort
One command plus a key — uvx mcp-server-whisper, then supply credentials
USE CASES
Transcribing the meeting you just recorded
Grab the most recent file and transcribe it without hunting for the path.
Finding a recording you only half remember
Filter the directory by length and roughly when it was made, then work from what comes back.
Turning an interview into something readable
Transcribe with the template that keeps tone and background, or the one that produces a formal write-up.
Asking a question about a long recording
Convert it to a supported format and query the audio directly instead of reading the whole transcript.
A recording too large to submit
Compress it first, then transcribe.
Producing narration from a script
Generate speech with the voice, delivery instruction and speed you want.
8 tools
Listed in project docs. Your client may show a different list.
get_latest_audioconvert_audiocompress_audiotranscribe_audiochat_with_audiotranscribe_with_enhancementcreate_audioBEFORE YOU INSTALL
What to know first
Tools you get
8 tools named in the project's documentation.
Requires
OPENAI_API_KEY in your environment
Where it runs
Both documented setups run the server on your machine.
INSTALLATION
Connect your client
Server setup
uvx mcp-server-whisperMCP Clients
Claude Desktop — uvx
{
"mcpServers": {
"whisper": {
"command": "uvx",
"args": [
"mcp-server-whisper"
],
"env": {
"OPENAI_API_KEY": "<YOUR_OPENAI_API_KEY>",
"AUDIO_FILES_PATH": "<YOUR_AUDIO_FILES_PATH>"
}
}
}
}Claude Desktop — local build
{
"mcpServers": {
"whisper": {
"command": "uv",
"args": [
"run",
"mcp-server-whisper"
],
"env": {
"OPENAI_API_KEY": "<YOUR_OPENAI_API_KEY>",
"AUDIO_FILES_PATH": "<YOUR_AUDIO_FILES_PATH>"
}
}
}
}