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Normalize MP3 Loudness

Free Online MP3 Loudness Normalizer

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How to Normalize the Loudness of MP3

1 Upload the MP3 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 MP3 file, with the dynamics inside each track left untouched.

Normalize MP3 Loudness FAQ

Does the source format affect normalisation?
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It does. MP3 is the universal lossy codec — nothing refuses to play it, at the cost of poorer quality per bit than AAC or Opus. That determines whether the corrected audio can be written back exactly or has to be re-encoded.
Yes — MP3 frames are fixed-size blocks, so a cut lands on a frame boundary and gapless playback markers do not survive a re-encode. Worth knowing when the goal is a consistent set rather than one file.
Concretely, loudness is measured against the EBU R128 standard and a single gain adjustment brings the whole file to the target LUFS — which is what streaming platforms actually measure, and is not the same thing as peak normalisation. 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 metadata atoms are carried across, cover art included, so a processed file still sorts correctly in a music library. ReplayGain values are recalculated rather than copied, because a stale one is worse than none.
MP3, AAC, M4A, OGG, Opus and WMA are the lossy codecs handled directly, and lossless sources (WAV, FLAC, AIFF) can go in the same batch. The codec is probed from the stream, so an incorrectly named file still works.
Yes: free accounts process audio up to 200 MB per file, which at typical bitrates is several hours of material; ffmpeg does the decoding and encoding, exactly as it would on a desktop. That is the most generous cap on the site after video, and audio work rarely runs into it.
A lossy file has already discarded detail once, and re-encoding discards more — that is generation loss and it does not undo. Wherever the job can be completed by copying the encoded stream rather than re-encoding it, that is what happens, and the audio comes back bit-identical.
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 MP3 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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