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Normalize AAC Loudness

EBU R128 AAC Leveller

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How to Normalize the Loudness of AAC

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

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.
No. It is a single gain adjustment for the whole file, so the relationship between the quiet and loud parts is untouched. Compression and limiting are separate operations and are not applied here.
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.
No. Normalize AAC Loudness is free without an account and adds no audio watermark or spoken tag to the output. Your upload is deleted from the workers shortly after the job completes.
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.
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.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.
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.
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.
No account, and nothing is kept: uploads are deleted from the workers shortly after the job finishes, nothing is listened to and nothing is indexed. Free accounts exist for history and batch size, not for access.

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