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Supported: MP3, WAV, FLAC, M4A, AAC, OGG, OPUS
Max file size: 200 MB
Normalization increases quiet audio and reduces loud audio to the target level. Prevent clipping caps the applied gain so the file stays within the true-peak ceiling.
Recommended: -14 (Streaming), -16 (Standard), -23 (Podcast)
Prevent clipping by limiting the true peak level.
Maintain balance between channels
Automatically adjust to prevent distortion
Your audio is processed in your browser. No files are uploaded.
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All processing happens in your browser. Your files never leave your device.
Audio Normalizer balances loudness across a recording so it lands at a consistent target level. Quiet parts come up, loud peaks are tamed, and the file stays under a true-peak ceiling so lossy encoders do not introduce inter-sample clipping. It is especially useful for podcasts that need to match streaming loudness norms, voice memos that vary between sessions, music libraries that play back at uneven volumes, and any voice-over work that needs to land on a known LUFS value before delivery.
Streaming platforms (Spotify, YouTube, Apple Music) rebroadcast all audio at a fixed perceived loudness. If a track is delivered too loud, the platform turns it down — losing dynamic range. If it is too quiet, the platform may boost it and reveal noise. Normalizing locally lets you control where the level lands and, just as important, what the true-peak ceiling is.
MP3 at 192–320 kbps is the most compatible delivery format. M4A (AAC in MP4) is best for the Apple ecosystem. Pick FLAC or WAV when you need a lossless master for further editing. OGG is good for open-source toolchains. Keep sample rate on Keep original unless a downstream tool requires a specific value — resampling does not improve audible quality of already-recorded audio.
No. Decoding, loudness analysis, normalization, and encoding all happen inside your browser using the Web Audio API and WebCodecs. Files never leave your device.
LUFS (Loudness Units relative to Full Scale) measures perceived loudness. Common targets: -14 LUFS for streaming platforms (Spotify, YouTube, Apple Music), -16 LUFS for general podcast/voice, -23 LUFS for broadcast (EBU R128).
True peak is the highest sample value, measured in dBTP. Setting -1.0 dBTP leaves 1 dB of headroom so lossy encoders (MP3, AAC) do not introduce inter-sample clipping. Use -2.0 dBTP for safer mastering.
EBU R128 targets perceived loudness via LUFS — the most accurate for mixed material. ReplayGain is similar but historically calibrated for music libraries. Peak Normalization scales the file so the loudest sample hits the target — fast but ignores perceived loudness.
It caps the applied gain so the resulting peak does not exceed Max True Peak. If a chosen LUFS target would force gain that pushes the file into clipping, Prevent Clipping reduces the gain to stay within the headroom.
Normalization is a static gain offset applied to the whole file. The displayed value is the difference (in dB) between the original loudness and the target. Negative means quieter, positive means louder.
We apply a BS.1770-style K-weighting filter (high-shelf + high-pass) before computing mean-square loudness. The output is within ~0.5 LUFS of professional meters for typical material — ideal for browser-side normalization, not for broadcast certification.
200 MB. Decoding, analysis, and rendering all stream through memory rather than buffering the whole file twice, so long voice recordings still process responsively.