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Normalize Audio Loudness

Free Online Audio Loudness Normalizer

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How to Normalize the Loudness of Audio

1 Upload the audio files you want brought to a consistent level — a whole set at once is exactly the case this is for.
2 Pick a target: about -14 LUFS for streaming platforms, -16 LUFS for podcasts, -23 LUFS for broadcast.
3 Run the analysis; integrated loudness and true peak are measured before any gain is decided on.
4 Download the normalized Audio file, with the dynamics inside each track left untouched.

Normalize Audio Loudness FAQ

Does the source format affect normalisation?
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It does. this format is read and written by the same streaming pipeline as everything else we support. That determines whether the corrected audio can be written back exactly or has to be re-encoded.
Yes — the codec is detected from the stream, so a mislabelled extension does not derail the job. Worth knowing when the goal is a consistent set rather than one file.
Concretely, the file is analysed for integrated loudness and true peak, then a single gain is applied to reach the target — no compression, no limiting, just the level. The file is measured first and the correction follows from the measurement, rather than a fixed gain being applied and hoped for.
A volume change applies a number you chose. Normalisation measures how loud the file actually is and computes the gain needed to hit a target, so ten files recorded at ten different levels all arrive at the same one.
Yes — ID3, Vorbis comments and MP4 atoms are carried across along with embedded artwork, so processed files still sort correctly in a music library.
MP3, WAV, AAC, M4A, FLAC, OGG, Opus, WMA and AIFF. Codecs are detected from the stream rather than the extension, so a mislabelled file still works and a mixed batch is handled per file.
Yes: free accounts process audio up to 200 MB per file whichever codec you brought; ffmpeg handles the decode and encode. In practice that limit only bites on uncompressed WAV; a compressed file of the same length is nowhere near it.
It depends on the source. Lossless input (WAV, FLAC, AIFF) is edited sample-exactly and loses nothing. Lossy input has already discarded detail once, so where the job can be done by copying the encoded stream rather than re-encoding it, that is the path taken.
WEBP.to is built around the modern replacement for both of the old defaults: one format that does lossy, lossless, alpha and animation, and reliably lands smaller than the JPEG or PNG it came from. People arrive here having decided the old defaults are costing them bandwidth, and that decision immediately raises the rest: what size, what quality, what happens to the transparency. Normalize Audio Loudness is on the same upload and the same account so those questions get answered in one place.
The converter on this site moves images to and from WebP, PNG, JPG and AVIF, which is where you decide whether a smaller file is worth losing the older browsers. Doing it in that order matters: pick the pixels first and the container afterwards, because the container is the cheap decision and the pixels are the expensive one.
The engines are identical — same libraries, same workers, same caps. What is different is the opinion attached, and the opinion here rests on a property of the format this site is named after: the same .webp extension can hold a lossy VP8 frame, a lossless one, or an animation, so the sensible processing path is decided per file rather than per extension.
No account, and nothing is retained: uploads are deleted from the workers shortly after the job finishes, and nothing is inspected, listed or indexed. Accounts exist for history and batch size only.

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