Live Audio Intelligence

io.github.ykshah1309Published by GitHub account ykshah1309v0.1.3Updated Oct 24, 2026

Live webcast transcription + vocal stress analysis (F0 jitter, hesitation) for earnings calls.

VerifiedSTDIODesktop onlyFinanceData & AnalyticsMedia & Design

Overview

AI-generated overview

Transcribes live financial webcasts in real time and returns a heuristic vocal stress score for the speaker.

What you can do
Resolves a live webcast URL, pulls the audio with ffmpeg, and transcribes it in rolling chunks using faster-whisper on CPU. Exposes four tools: monitor_live_stream to start a stream and return a stream_id, get_rolling_transcript for the last N minutes of text, analyze_speaker_stress for a 0-100 stress score with pitch jitter, hesitation ratio and pause stats, and stop_monitor to tear the stream down. The stress score is a hand-weighted composite of prosodic features and is explicitly described as heuristic, not validated against market outcomes.
When to use it
Useful when an assistant needs live transcription of earnings calls, investor days or business news streams, plus a coarse read on how stressed the speaker sounds. Suited to analysts who want to ask about what was just said and how it was said in the same conversation.
Requirements
Local process over stdio; Python 3.10 or newer, installed from PyPI or run with uvx. FFmpeg must be installed separately as a system binary and be on PATH. First run downloads the faster-whisper base.en model (about 140 MB) from Hugging Face, so network access is needed. Optional environment variable LAI_MAX_CONCURRENT_STREAMS raises the default cap of 4 concurrent streams.
Before you install
The server fetches and processes third-party webcast audio, so stream URLs and their content pass through the local pipeline. It spawns ffmpeg and yt-dlp subprocesses and writes temporary files. The stress score is a heuristic with hand-picked weights and saturation points, not empirically validated against market outcomes, so it should not be treated as a trading signal. No credentials are declared.

Installation

In SourceWeft

  1. Open Live Audio Intelligence in the dashboard and add it to a workspace.
  2. Enable the server for the chats that should use its tools.

Desktop only via STDIO. STDIO servers start a local process, so they need the SourceWeft desktop host.

Other MCP clients

Follow the launch instructions in the repository.

README

live-audio-intelligence-mcp

MCP server for live financial webcast transcription and heuristic vocal stress analysis.

Turns any live webcast URL (earnings calls, CNBC, investor days) into a real-time pipeline that feeds an LLM two things simultaneously:

  1. A rolling transcript via faster-whisper (CPU, int8).
  2. A heuristic vocal stress score (0–100) derived from F0 pitch jitter, hesitation ratio, and voiced-frame fraction. These prosodic features are well-established correlates of speaker arousal in the vocal-analysis literature; their composition into the score below is heuristic and has not been empirically validated against market outcomes. Treat it as a coarse signal, not an oracle.

Built on the Model Context Protocol. Exposes 4 tools over stdio; drop it into Claude Desktop, Claude Code, or any MCP client.

Why this exists

Sell-side analysts and hedge-fund PMs don't just want to read the earnings transcript after the fact — they want a real-time signal about how confident the CFO sounds when asked about Q4 guidance. This server wires a Whisper pipeline and a pYIN-based prosody analyzer directly into an LLM's tool loop, so the model can ask "what did the CEO just say about China?" and "how stressed did they sound saying it?" in the same conversation.

Install

1. System prerequisite — FFmpeg

FFmpeg is a system binary, not a Python package. The ffmpeg-python wrapper is not a dependency here — we drive the binary directly via subprocess. You must install it yourself.

macOS (Homebrew):

bash
brew install ffmpeg

Linux (Debian / Ubuntu):

bash
sudo apt-get update && sudo apt-get install -y ffmpeg

Linux (Fedora / RHEL):

bash
sudo dnf install -y ffmpeg

Windows — choose one:

powershell
# Option A — winget (Windows 10/11)winget install --id=Gyan.FFmpeg -e
# Option B — Chocolateychoco install ffmpeg
# Option C — Scoopscoop install ffmpeg

Confirm it's on your PATH:

bash
ffmpeg -version

If the command errors with "not found", reopen the terminal (PATH changes don't propagate to already-open shells) or add the ffmpeg bin/ directory to your PATH manually.

2. Python package

Requires Python ≥ 3.10.

bash
pip install live-audio-intelligence-mcp

Or run directly without installing with uv:

bash
uvx live-audio-intelligence-mcp

The first run will download the faster-whisper base.en model (~140 MB) from Hugging Face and cache it under ~/.cache/huggingface/.

Run it

Stdio MCP server:

bash
live-audio-intelligence-mcp

Or equivalently:

bash
python -m live_audio_intelligence_mcp

Claude Desktop

Add to claude_desktop_config.json:

json
{  "mcpServers": {    "live-audio-intelligence": {      "command": "live-audio-intelligence-mcp"    }  }}

Claude Code

bash
claude mcp add live-audio-intelligence -- live-audio-intelligence-mcp

Tools

ToolPurpose
monitor_live_stream(url, disable_vad=False)Resolve the audio URL, spawn ffmpeg, start chunking + transcription. Returns a stream_id.
get_rolling_transcript(stream_id, minutes_back=10)Get the last N minutes of concatenated transcript text.
analyze_speaker_stress(stream_id, time_window_seconds=60)Run prosody analysis over the last N seconds of audio. Returns stress score, pitch jitter, hesitation ratio, pause stats, and a human-readable interpretation.
stop_monitor(stream_id)Kill ffmpeg, clean up temp files, drop the transcript buffer.

The stress score

ScoreInterpretation
0–20Confident, fluent delivery
20–45Normal variation
45–75Elevated stress — worth monitoring
75–100High stress — potential market-moving signal

Composite of:

  • Pitch jitter (coefficient of variation of F0) — 50% weight, saturating at jitter = 0.12
  • Hesitation ratio (fraction of audio in pauses > 400 ms) — 35% weight, saturating at 0.30
  • Unvoiced fraction (speaker trailing off) — 15% weight

The three features are literature-backed correlates of speaker arousal (see pYIN for F0 tracking, and the broad "disfluency is a correlate of cognitive load" line of work). The weights and saturation points are hand-picked defaults, chosen so that a calm speaker scores in the 0–20 band on clean studio audio and visibly stressed speech scores ≥ 45 — they are not fit to any labeled dataset. Consumers who care about absolute numbers should recalibrate thresholds against their own recordings.

A synthetic-audio calibration harness lives at scripts/validate_stress_score.py. It generates controlled audio (smooth sine, jittered pitch, silence-padded speech) and asserts that the score responds in the expected direction. This is calibration evidence, not market-outcome validation.

Low-SNR mode

For speakerphone audio (most earnings Q&A), pass disable_vad=true to monitor_live_stream. Silero VAD tends to aggressively classify muddy conference-call speech as silence; disabling it preserves more of the speech at the cost of transcribing a bit more ambient noise.

Concurrency limits

By default the server caps concurrent streams at 4 (each stream holds an ffmpeg subprocess, a yt-dlp subprocess, a thread, and a temp directory). Override via env var for high-throughput deployments:

bash
LAI_MAX_CONCURRENT_STREAMS=16 live-audio-intelligence-mcp

Exceeding the cap raises StreamLimitExceededError rather than silently queuing.

Architecture

                 ┌──────────────────┐    URL  ─────▶  │  yt-dlp resolve  │                 └────────┬─────────┘                          │ audio URL                          ▼                 ┌──────────────────┐      ┌────────────────┐                 │  ffmpeg (bg)     │ ───▶ │  15s WAV chunk │                 │  16kHz mono PCM  │      │  queue         │                 └──────────────────┘      └───────┬────────┘                                                   │                                ┌──────────────────┴────────────────┐                                ▼                                   ▼                       ┌──────────────────┐              ┌──────────────────┐                       │ faster-whisper   │              │  librosa.pyin    │                       │ (int8 / CPU)     │              │  + pause detect  │                       └────────┬─────────┘              └────────┬─────────┘                                │ rolling transcript              │ stress score                                ▼                                 ▼                            ┌────────────── MCP stdio ───────────────┐                            │    LLM (Claude) — calls tools freely   │                            └────────────────────────────────────────┘

All blocking work (Whisper inference, ffmpeg I/O, librosa DSP) is dispatched to threads via asyncio.to_thread so the MCP event loop stays responsive.

Development

bash
git clone https://github.com/ykshah1309/live-audio-intelligence-mcpcd live-audio-intelligence-mcppython -m venv .venvsource .venv/bin/activate  # Windows: .venv\Scripts\activatepip install -e ".[dev]"pytestlive-audio-intelligence-mcp

Running the tests

The pytest suite in tests/ covers the pure-Python logic that doesn't require network or ffmpeg:

  • URL syntactic validation (scheme allow-list, host presence)
  • Concurrency-cap enforcement in StreamManager
  • Custom exception hierarchy (backward-compat with ValueError / RuntimeError)
  • Prosody analyzer on synthetic audio (sine tone, silence, jittered pitch)
bash
pytest -q

Calibration benchmark

bash
python scripts/validate_stress_score.py

This generates synthetic audio with known acoustic properties and verifies the stress score responds in the expected direction. It's a sanity check for the weighting heuristics — not a replacement for empirical validation against real earnings-call outcomes.

Troubleshooting

ffmpeg: command not found — ffmpeg isn't on PATH. See the install section above. On Windows, reopen your terminal after installing.

yt-dlp could not resolve URL — The site isn't supported by yt-dlp or the URL is malformed. Test with yt-dlp -F <url> from the command line; if that fails, the server will too.

Whisper downloads hang on first run — The ~140 MB model download goes to ~/.cache/huggingface/. Check your network and Hugging Face access.

"Insufficient voiced frames" in stress output — The audio window is mostly silence or noise. Usually means the stream is still buffering; wait 30s and retry. For speakerphone Q&A, start the monitor with disable_vad=true.

Contributing

See CONTRIBUTING.md.

Changelog

See CHANGELOG.md.

License

MIT — see LICENSE.

Source: README.md at commit 71f03a3

Tools

0
SourceWeft has not started this package in its sandbox yet, so its tools are not known.

Version history

1
  1. v0.1.3LatestSep 16, 2026

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