voice-analysis-mcp
Provides local audio analysis tools for LLMs, enabling transcription, conversation dynamics, prosody analysis, and visual inspection without API keys.
README
voice-analysis-mcp
An MCP server that gives LLMs like Claude the tools to review and do qualitative analysis of call recordings, voice memos, TTS output, and other audio files — transcription, conversation dynamics, vocal delivery, recording quality, and visual inspection via spectrograms/waveforms.
Everything runs locally: ffmpeg for decoding, faster-whisper for transcription (models download on first use), librosa for signal analysis. No API keys required.
Tools
| Tool | What it answers |
|---|---|
get_audio_info |
What is this file? Duration, channels, sample rate, codec, tags. |
transcribe |
What was said, when? Timestamped segments, optional word timing, per-channel (per-speaker) transcription for stereo calls, words-per-minute. |
analyze_conversation |
Who talks how much? Talk time and turns per channel, dead air, overlap, interruption counts, response latency (stereo calls: one speaker per channel). |
analyze_prosody |
How does it sound? Pitch median/range (monotone vs expressive), loudness and dynamic range, pace, pause/hesitation patterns for a window (≤ 300 s). |
analyze_quality |
Can I trust this recording? Levels, clipping, noise floor, estimated SNR, narrowband detection, flagged issues. |
extract_segment |
Cut a key moment out to a standalone wav file. |
render_spectrogram |
A mel spectrogram image the model can look at — spot hold music, DTMF/beeps, hum, dropouts, TTS artifacts. |
render_waveform |
Per-channel waveform image — speaker activity, silences, level imbalance at a glance. |
A typical call review: get_audio_info → analyze_conversation (whole file,
cheap) → transcribe (per channel for stereo) → analyze_prosody /
render_spectrogram on the interesting windows.
Requirements
- Python 3.11–3.13 and uv
- ffmpeg on PATH (
brew install ffmpeg)
Setup
uv sync
uv run pytest # optional: verify
Claude Code
Working inside this repo, nothing to configure: the checked-in .mcp.json
registers the server automatically (approve it when prompted). From any other
project, register it globally:
claude mcp add --scope user voice-analysis -- uv run --directory /path/to/voice-analysis.mcp voice-analysis-mcp
Claude Desktop (claude_desktop_config.json)
{
"mcpServers": {
"voice-analysis": {
"command": "uv",
"args": ["run", "--directory", "/path/to/voice-analysis.mcp", "voice-analysis-mcp"]
}
}
}
How a Claude instance uses this
The model only needs a file path — every tool takes an absolute path to a local audio file, so "analyze the call in ~/Downloads/call-4711.mp3" is enough to start. The server's instructions steer the model through the intended flow:
get_audio_info— learn duration and channel count (drives everything else)analyze_conversation— cheap whole-file pass: talk balance, dead air, interruptionstranscribe— content, per channel on stereo calls for who-said-whatanalyze_prosody/analyze_quality/render_spectrogramon windows the first passes flagged — all results use absolute seconds, so findings from different tools line up
Example prompts that work end-to-end:
Review ~/calls/support-0412.wav: who dominated the conversation, were there awkward silences, and how did the agent's tone change after the customer pushed back?
Compare the TTS output in v1.wav and v2.wav — which sounds less monotone and are there any audible artifacts?
Results are compact JSON (rounded numbers, capped lists) sized for a model's context window; images come back as MCP image content Claude can view directly.
Notes & design choices
- Stereo = per-speaker. Telephony platforms typically export calls with one
speaker per channel. Tools take a
channelparameter to isolate a speaker;analyze_conversationuses both channels for interruption/overlap/latency metrics. Mono recordings fall back to overall speech/silence dynamics — model-based diarization (e.g. pyannote) is a possible future addition. - Whisper models:
tiny/base/small/medium/large-v3/large-v3-turbo, cached under the Hugging Face cache dir after first download.baseis the default trade-off; usesmall+ for noisy telephone audio. - Windows over whole files for the expensive analyses: prosody is capped at
300 s and images at 600 s per call — pass
start_time/end_time. All timestamps in results are absolute seconds into the file, so findings from different tools line up. - Any input format works if ffmpeg can decode it, including pulling the audio track out of video files.
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