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About this tool
Normalize audio loudness with LUFS, true-peak and clipping analysis.
Loudness & True-Peak Normalizer runs the EBU R128 loudness normalisation pass (FFmpeg's loudnorm filter) over an audio or video file, retargeting integrated loudness to -14, -16 or -23 LUFS while holding true peak at -1.5 dBTP and loudness range at 11 LU. Unlike peak normalisation, which just scales the waveform until the loudest sample hits 0 dBFS, this measures perceived loudness across the whole programme, so two files normalised to the same LUFS target actually sound equally loud. Podcasters, video editors and musicians get a WAV that meets the platform spec instead of guessing at a fader move.
Open Loudness & True-Peak Normalizer on AltFTool — it loads instantly in your browser.
Pick an audio or video file under Source file — the picker accepts audio/* and video/* input.
Set Target to -14 LUFS, -16 LUFS or -23 LUFS and press Process locally; FFmpeg WebAssembly runs loudnorm=I=target:TP=-1.5:LRA=11 on the file in memory.
The normalised audio downloads as altftool-loudness-true-peak-normalizer.wav, with the FFmpeg log shown in the Local processing report.
Integrated LUFS measurement matches how loudness is perceived over time, so a quiet track with one loud transient is not left quiet the way peak normalisation leaves it.
Every target is rendered with a -1.5 dBTP ceiling, which is the margin that keeps reconstructed inter-sample peaks from clipping after MP3 or AAC encoding.
You can point it at a video file and get the normalised audio out, without demuxing the track by hand first.
-16 LUFS integrated for stereo is the widely used podcast target and matches Apple Podcasts' recommendation; mono spoken word is usually delivered around -19 LUFS. Pick -16 LUFS here and keep the true peak at or below -1 dBTP.
dBFS measures sample amplitude at a single instant, while LUFS measures perceived loudness averaged over the programme using the R128 frequency weighting. That is why two files can both peak at 0 dBFS and still differ by 8 LU in how loud they sound.
Because lossy encoders reconstruct a waveform that can overshoot the original samples. EBU R128 permits a maximum of -1 dBTP, and -1.5 dBTP gives an extra half decibel of margin so MP3, AAC and Opus versions do not clip on playback.
-14 LUFS is the usual streaming reference — Spotify normalises playback to about -14 LUFS by default. Mastering much louder than that gains you nothing on those platforms, because the service simply turns your track back down.