Audio Normalizer

What is Audio Normalizer?

A loudness normalizer that makes clips from different microphones, rooms and recorders sound equally loud when played back to back.

How it works

The current level is measured locally from the decoded samples (peak, RMS and an estimated LUFS). The export then applies ffmpeg's loudnorm filter — the same EBU R128 algorithm broadcasters use — or dynamic normalisation with a hard limiter in peak mode.

1. Load your file. Drag a file onto the drop zone or pick one from your device. It is read locally by the browser and never uploaded.
2. Set the options. Adjust the handles, format, quality or threshold controls — the preview updates as you change them.
3. Process and download. Run the tool, then download the result. Compare the reported before/after size for compressors and converters.

Examples

A 42 MB podcast episode → 9 MB MP3

Trim the intro, normalise to −16 LUFS, and export at 96 kbps mono — a typical interview lands around a fifth of the original size.

A phone video with the wrong orientation

Rotate 90° and the width/height swap in the output, so it plays upright in every player instead of relying on metadata flags.

Common mistakes

Expecting instant encoders

Video encoding runs in single-threaded WebAssembly, so a clip takes roughly its own length (or a little longer) to re-encode. Stream-copy operations are near-instant by comparison.

Re-encoding when a copy would do

Trimming, removing audio and joining same-codec clips can copy streams without re-encoding — pick the copy option when it is offered to avoid any quality loss.

Ignoring the first-load download

The media engine is about 31 MB and is fetched on first use, then cached. The first conversion on a connection is slower than every later one.

Frequently asked questions