Audio File Info & Quality Analyzer

One upload, a full technical readout: duration, sample rate, channels, approximate bitrate, peak level, estimated loudness, DC offset, and a clipping check — genuinely measured from your file, not guessed.

How to use the Audio File Info & Quality Analyzer

  1. Add your audioDrop a file onto the page or tap to browse — or press “try a built-in sample” to explore first. Nothing is uploaded; the file is decoded on your device.
  2. Analysis runs automaticallyA full report appears: format facts, loudness, peaks, clipping, DC offset, stereo health and an effective-bandwidth check.
  3. Verify and use the resultUse the buttons under the report to listen and check; copy the result with one click.

About the Audio File Info & Quality Analyzer

The Audio File Info Analyzer tells you everything measurable about an audio file: duration, true sample rate, channels, average bitrate, loudness (LUFS), loudness range, true peak, RMS, crest factor, clipping, DC offset, silence share, stereo correlation and width — and whether a ‘lossless’ file is really an upsampled MP3.

That last check is unique: lossy encoders low-pass the audio at a fixed frequency (about 16 kHz for 128 kbps MP3, 19–20 kHz for 192–320 kbps). A genuine lossless recording has content right up to the Nyquist limit, so a sharp cliff in its spectrum reveals a transcode.

How the Audio File Info & Quality Analyzer works

The browser decodes the file at its true sample rate (read from the header, so 96 kHz files stay 96 kHz). Loudness and true peak follow ITU-R BS.1770-4. Effective bandwidth is found from the long-term average spectrum (4096-point FFT over up to 60 s) by locating the steepest drop of at least 18 dB over about 400 Hz above 8 kHz; stereo correlation is the Pearson coefficient of left and right; clipping counts runs of two or more samples at full scale; silence is the share of 50 ms blocks below −60 dBFS.

What people use it for

Authenticity check

Detect MP3s masquerading as FLAC/WAV.

QC before delivery

Verify levels, clipping and format.

Podcast checks

Confirm loudness and stereo health.

Debugging

Find out why a file sounds wrong.

Tips for the best results

  • A bandwidth near 16 kHz in a WAV or FLAC almost always means it was converted from a 128 kbps MP3.
  • A correlation below 0 means the file will thin out in mono.
  • Clipped samples can’t be repaired — re-export from the source at lower gain.

Common problems & fixes

Bitrate looks oddIt is file size ÷ duration, so tags and cover art add a little.
No bandwidth cliff foundThe file is full-band or naturally rolled off; this is normal for genuine high-quality audio.

Quick facts

ReadsAny browser-decodable file
Sample rateRead from the header, not resampled
Metrics13–16 including stereo health
Transcode checkSpectral cliff detection

Frequently asked questions

How can you tell a fake FLAC?

By its spectrum: lossy-sourced FLACs end at a lossy encoder’s low-pass frequency.

What is crest factor?

Peak level minus RMS level; low values mean heavy compression.

What does 'clipped samples' mean?

It's a count of samples that hit (or came within a hair of) full digital scale — a high count usually means the recording or a processing step pushed the level too hot and introduced audible distortion.

Is the LUFS reading exact?

It's a fast RMS-based estimate, not a certified ITU-R BS.1770 loudness meter — good for a quick reference, not for meeting a broadcast/streaming loudness spec exactly.

Why does 'approximate bitrate' say approximate?

It's back-calculated from file size and duration (size × 8 ÷ duration), which is exact for constant-bitrate files but only an average for variable-bitrate files.

Your audio never leaves your device

This tool runs entirely in your browser using the Web Audio API. Nothing is uploaded, stored or shared — there is no server to send your file to, so it works offline once the page has loaded.

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