Audio RMS level report

About this tool

Use this tool when you need a whole-file level check before publishing, normalizing, matching stems, or debugging clipped audio. Paste a WAV, FLAC, MP3, OGG/Vorbis, M4A/MP4, AIFF, CAF, MKV/WebM, or AAC-ADTS file as base64 or hex and it reports each decoded channel plus an overall row.

The JSON summary includes sample rate, channel labels, duration, frame counts, whole-file RMS, sample peak, mean absolute level, short-window RMS peak/trough, crest factor, headroom, DC offset, zero crossings, and clipped-sample counts. Choose CSV for spreadsheets or the text report for a quick terminal-style readout.

Worked example

For the preset constant half-scale WAV, JSON output includes values like:

{
  "sample_rate": 8000,
  "channels": 1,
  "duration_s": 0.01,
  "per_channel": [
    {"channel": 1, "label": "M", "rms_dbfs": -6.021, "peak_dbfs": -6.021}
  ]
}

The same input in CSV mode starts with:

channel,label,samples,rms_dbfs,peak_dbfs,average_dbfs
1,M,80,-6.021,-6.021,-6.021

Limits and edge cases

• Input must be pasted as base64 or hex file bytes; browser file upload is not used for this pure text surface.
• The decoded input byte cap is 64 MiB and analysis stops after 30,000,000 frames per channel, reporting truncated: true if that cap is reached.
• Opus, AC-3, DTS, encrypted media, and video-only files are not decoded by the pure-Rust backend.
• RMS and peak are sample-based dBFS values, not LUFS or inter-sample true peak. Use a loudness-meter tool when broadcast LUFS or true-peak compliance is required.
• The short RMS window controls only the rms_peak_dbfs and rms_trough_dbfs fields; it does not change whole-file RMS, average, or sample peak.

FAQ