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audio-quality-check

@tenequm · 收录于 1 周前

Analyze audio recording quality - echo detection, loudness, speech intelligibility, SNR, spectral analysis. Use when the user wants to check a recording's quality, detect echo or duplication in audio files, measure speech clarity, compare original vs processed audio, diagnose why a recording sounds bad, or analyze audio tracks from Blackbox or any call recording app. Triggers on audio quality, recording analysis, echo detection, check recording, sound quality, analyze audio, speech quality, PESQ, STOI, loudness, SNR, audio diagnostics, recording sounds bad, echo in recording, audio duplication.

适合你,如果经常需要检查录音文件的质量问题

/ 下载安装
audio-quality-check.skill双击,或拖进 Claude 桌面版 / Cowork,即完成安装↓ .skill↓ .zip
用别的 agent?下载 .zip 解压,把文件夹放进它的技能目录
Claude Code~/.claude/skills/(项目级 .claude/skills/)
Codex CLI~/.codex/skills/
Cursor自动读取上面两处目录
其他工具见其文档的「skills」目录;两个下载是同一份文件,只是名字不同
/ 通过 npx 安装 校验哈希
npx oh-my-skill add tenequm/skills/audio-quality-check
/ 通过 bash 安装
curl -fsSL https://oh-my-skill.com/install.sh | bash -s -- tenequm/skills/audio-quality-check
/ 已经装过?验证本机副本,不用重装
npx oh-my-skill verify tenequm/skills/audio-quality-check
安装目标可用 --agent / --scope 或 --to 明确指定;省略时只会在唯一已存在的 agent 目录上自动选择,零命中或多命中会停止并提示。content_hash 缺失或不一致均拒装。
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怎么用

技能原文 SKILL.md作者撰写 · Apache-2.0 · ecb77f8

Audio Recording Quality Analyzer

Comprehensive audio quality analysis for call recordings. Handles dual-track M4A files (system audio + mic), single-track recordings, and AEC-processed files.

Quick Start

Run the bundled analysis script on a recording directory:

python <skill-path>/scripts/analyze_recording.py "/path/to/recording/directory"

Modes for focused analysis:

python <skill-path>/scripts/analyze_recording.py /path --tracks   # track info only
python <skill-path>/scripts/analyze_recording.py /path --echo     # echo detection only
python <skill-path>/scripts/analyze_recording.py /path --quality  # quality metrics (skip echo)

For Blackbox recordings, the directory is typically: ~/Library/Application Support/Blackbox/Recordings/<timestamp-id>/

Dependencies

System: ffmpeg, ffprobe (brew install ffmpeg) Python: numpy, soundfile, scipy, pyloudnorm, pesq, pystoi, librosa

Install all Python deps: pip3 install numpy soundfile scipy pyloudnorm pesq pystoi librosa

What Each Metric Tells You
EBU R128 Loudness (pyloudnorm)
  • What: Perceptual loudness in LUFS (Loudness Units Full Scale)
  • Target: -16 to -24 LUFS for speech
  • Watch for: AEC/post-processed tracks being significantly louder than originals (indicates the processing is amplifying without normalizing)
Echo Detection - Autocorrelation
  • What: Detects delayed copies of the signal within a single track by correlating the signal with itself at various time offsets
  • How to read: Peaks in the 20-100ms range with correlation > 0.3 indicate signal duplication. The lag tells you the delay of the duplicate copy
  • Key insight: If you see a consistent peak at the same lag across multiple time segments, that's a systematic duplication (e.g., a virtual audio processor like Krisp introducing a delayed copy at ~53ms)
  • Normal values: Peaks below 0.15 are typically speech pitch harmonics (harmless). Peaks above 0.3 at consistent lags are echo
Cross-Track Correlation
  • What: Measures how much one track's content appears in another (e.g., system audio bleeding into the mic track)
  • How to read: Values near 0 mean no bleed. Values above 0.1 indicate the mic is picking up system audio
  • Coherence: Frequency-domain version of the same test. Voice-band coherence (300-3400Hz) is most relevant for speech echo
PESQ - Speech Quality (requires reference + degraded)
  • What: ITU-T P.862 standard. Gives a MOS (Mean Opinion Score) comparing a degraded signal against a reference
  • Scale: 1.0 (bad) to 4.5 (excellent). NB = narrowband (phone quality), WB = wideband
  • Use for: Comparing AEC-processed mic vs original mic to see if processing helps or hurts
  • Thresholds: 4.0+ excellent, 3.0+ good, 2.5-3.0 fair, <2.5 poor
STOI - Speech Intelligibility (requires reference + degraded)
  • What: Short-Time Objective Intelligibility. Measures how understandable speech remains after processing
  • Scale: 0.0 to 1.0
  • Thresholds: >0.8 good, >0.6 fair, <0.6 poor
  • Key insight: If STOI drops significantly between original and processed, the processing is degrading intelligibility
Spectral Analysis (librosa)
  • Centroid: Average frequency weighted by amplitude. Higher = brighter/harsher audio
  • Rolloff (85%): Frequency below which 85% of spectral energy sits. Lower = more bass-heavy
  • Zero-crossing rate: How often the signal crosses zero. Higher = noisier signal. Speech is typically 0.05-0.20; values above 0.30 suggest significant noise
SNR - Signal-to-Noise Ratio
  • What: Ratio of speech energy to background noise energy (estimated via energy-based VAD)
  • Thresholds: >20dB excellent, >15dB good, >10dB fair, <10dB poor
  • Note: This measures background noise, not echo. A recording can have excellent SNR but still have echo problems
Per-Minute Energy
  • What: RMS energy and voice-band energy per minute of recording
  • Use for: Spotting segments that went silent (mic cut out), got unexpectedly loud (clipping risk), or had activity patterns that help identify when speakers were active
Manual Analysis Recipes

When you need analysis beyond what the script provides, these patterns are useful.

Extract individual tracks from dual-track M4A
ffmpeg -y -i audio.m4a -map 0:0 -ac 1 -ar 16000 /tmp/system.wav
ffmpeg -y -i audio.m4a -map 0:1 -ac 1 -ar 16000 /tmp/mic.wav
Quick loudness check with sox
sox audio.wav -n stat 2>&1
Check specific time range for echo (Python)
import numpy as np
import soundfile as sf
from scipy import signal

data, sr = sf.read('/tmp/system.wav')
# Analyze 5 seconds starting at 2 minutes
start = 120 * sr
seg = data[start:start + 5*sr]
seg_norm = seg / (np.max(np.abs(seg)) + 1e-10)
autocorr = np.correlate(seg_norm, seg_norm, mode='full')
mid = len(seg_norm) - 1
autocorr = autocorr / autocorr[mid]
# Check 20-100ms range for echo peaks
min_lag = int(0.020 * sr)
max_lag = int(0.100 * sr)
region = autocorr[mid + min_lag:mid + max_lag]
peaks, props = signal.find_peaks(region, height=0.1)
for i, p in enumerate(peaks[:5]):
    lag_ms = (p + min_lag) / sr * 1000
    print(f"  Peak at {lag_ms:.1f}ms, r={props['peak_heights'][i]:.3f}")
Common Issues and What Causes Them

| Symptom | Likely cause | What to check | |---------|-------------|---------------| | Speakers sound slightly doubled/echoed | Virtual audio processor (Krisp) creating delayed copy in system audio | Autocorrelation: consistent peak at 40-60ms | | Mic track has remote speakers' voices | Acoustic echo (speakers to mic) | Cross-track correlation > 0.1 | | AEC-processed file sounds worse | DTLN-aec degrading signal quality | PESQ/STOI comparing original vs processed | | AEC-processed file is too loud | Missing loudness normalization after processing | Loudness: processed > -10 LUFS | | Recording has hiss/noise | Low SNR, noisy mic, or AGC artifacts | SNR < 15dB, high zero-crossing rate | | Quiet segments mid-recording | Mic cut out or device changed | Per-minute energy: sudden RMS drop |

按 Apache-2.0 许可原样转载,未经改动 · 在 GitHub 查看 →

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