1Upload the AAC audio files you want brought to a consistent loudness.
2There is nothing to set: the loudness is brought to -16 LUFS (EBU R128), with true peaks kept below -1.5 dBTP.
3Run the analysis; integrated loudness and true peak are measured before any gain is decided on.
4Download the normalized AAC file.
Normalize AAC Loudness FAQ
Anything specific to AAC here?
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Yes — AAC carries encoder delay in the container, so a naive cut can leave a few milliseconds of silence that a proper trim removes. Worth knowing when the goal is a consistent set rather than one file.
How does Normalize AAC Loudness decide the right level?
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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.
What is the difference between this and just turning it up?
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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.
Does Normalize AAC Loudness cost anything?
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No. Normalize AAC Loudness is free without an account, and nothing is added to the audio. The upload and the result are deleted automatically about 24 hours after the job, which leaves time to download the result.
What can I upload to Normalize AAC Loudness?
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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.
Is there a file size limit on Normalize AAC Loudness?
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Yes: free accounts process audio up to 15 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.
Why does a MP3 site host Normalize AAC Loudness?
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MP3.to is built around the format that made portable audio ordinary: universally playable, small enough to ignore, and lossy in a way that only matters if you keep re-encoding it. Audio jobs cluster: the same person who needs a format changed usually needs the file cut, levelled or made smaller in the same sitting. Putting Normalize AAC Loudness behind the same upload, the same caps and the same account is what turns three tabs into one.
What should I do with the result once Normalize AAC Loudness is finished?
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The converter on this site moves audio between MP3, WAV, FLAC, M4A, OGG and Opus, so the output format is a decision you can leave until the end rather than commit to at the start. Doing that after Normalize AAC Loudness rather than before means the encode happens once, on the file you actually settled on, instead of twice.
Is Normalize AAC Loudness here the same tool the sibling sites run?
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The engines are shared — the same ffmpeg build, the same workers, the same limits. What a MP3 site adds is a point of view about bitrates, about what survives an encode and about when a lossless intermediate is worth the upload time. It also starts from one fact about the format this site is named after: MP3 frames are fixed-size blocks, so a cut lands on a frame boundary and gapless playback markers do not survive a re-encode.
Do I need an account, and does anything get kept?
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No account is needed. Uploads and results are kept for about 24 hours, which leaves time to download the result, and are then deleted automatically; nobody reads, lists or indexes them. Several files in one job need a paid plan.
