Volume Normalizer
Normalize any track to a target loudness in LUFS with a safe true-peak ceiling, compare before and after, then export a WAV. Everything runs in your browser.
Drop a track to normalize
or choose an audio file
MP3 · WAV · FLAC · AAC · M4A · MP4 · OGG · OPUS · up to 25 MB
Want the full measurement first? Open the Loudness Analyzer.
Your audio stays on your device
This tool decodes, analyses and exports your track entirely in your browser. The file is never uploaded to our servers, and nothing is stored when you close the tab.
- Free
- No account
- Browser processing
How the volume normalizer works
The tool measures your track's integrated loudness using the ITU-R BS.1770-4 algorithm, calculates the difference between that measurement and your target, and applies that gain to every sample. Because it is a single, constant gain change, the relationship between the quietest and loudest moments of the music is preserved exactly.
The only complication is headroom. Raising a track by 4 dB also raises its peaks by 4 dB, and if those peaks were already close to full scale the result would clip. That is what the two modes exist to resolve, and the tool reports the gain it actually applied along with the before and after measurements, so you can see precisely what happened rather than trusting a progress bar.
Choosing a target
Use -14 LUFS for Spotify, YouTube and most streaming platforms, -16 LUFS if you are optimising for Apple Music's Sound Check, and -23 LUFS for European broadcast delivery under EBU R128. Podcasts commonly sit at -16 LUFS for stereo and -19 LUFS for mono.
For a batch of tracks that need to sit at a consistent level — a DJ folder, a playlist, a set of interview recordings — the target matters less than applying the same one to everything. Consistency is the point.
- -14 LUFS: Spotify, YouTube, Amazon Music, Tidal
- -16 LUFS: Apple Music Sound Check, stereo podcasts
- -19 LUFS: mono spoken-word podcasts
- -23 LUFS: EBU R128 broadcast delivery
Peak ceilings and why -1 dBTP matters
A digital file can be reconstructed into an analogue waveform that peaks higher than any stored sample. Lossy encoders make this worse: an MP3 or AAC derived from a file that already touches 0 dBFS routinely overshoots, and that overshoot becomes audible distortion on playback.
Leaving a ceiling of -1 dBTP costs you nothing perceptually and removes the problem entirely. The normalizer defaults to it, enforces it in both modes and never allows a sample past it in the exported file.
Frequently asked questions
It measures the perceived loudness of your track and applies a single gain change so the whole file lands on your chosen target. It does not re-compress or re-master the music: the balance between loud and quiet passages is preserved unless you deliberately switch on the limiter mode.