Audio Normalizer
Normalize audio by its loudest peak: set a ceiling anywhere from 1 to 100 percent, and a quiet take rises to meet it.
Runs entirely in your browser — your files and text never leave your device.
Drag & drop your file here, or
Normalizing raises the loudest peak to the chosen level (100% = 0 dBFS, the loudest without clipping).
Working on your device — keep this tab open until it finishes.
Bringing the level up
- Drop the quiet file on the Audio file box. It decodes and gets a player of its own underneath, which is worth thirty seconds of your attention, because whatever is loudest in there decides how far everything else can rise.
- Type a target peak. 100 leaves nothing spare, 89 keeps about a decibel back, and either way the figure applies to the file from start to finish.
- Pick M4A, WAV, Opus or FLAC, then press Normalize & download. A progress bar runs during the encode with Cancel beside it, and what comes back is a player plus a Download button, the file named after the one you loaded.
About this tool
A recording that sounds fine on your headphones and vanishes on someone else's phone usually has one thing wrong with it: its loudest moment sits well below full scale. Peak normalizing is the fix. The tool finds the single biggest sample in the file, works out what it would take to lift that one sample to your target, then multiplies every sample by the same number. Nothing about the balance of the recording shifts. All of it moves together.
Target peak is a percentage box that steps in whole numbers between 1 and 100. It starts at 100, which puts the loudest sample right at 0 dBFS. When you want headroom instead, the conversions are worth memorising: 89 is about a decibel down, 79 is two, 71 is three, 50 is a clean six. I leave it on 89 for anything heading back into an edit, since a file pinned at full scale has nowhere to go once the next tool adds gain.
Output goes to M4A, Opus, FLAC or WAV, and the WAV writer works at 16-bit, keeping whatever sample rate the source had. The other three get encoded from that same 16-bit copy, which makes this a good step for voice notes, lecture audio, podcast rough cuts and samples, and the wrong step for finishing a master. Matching several clips is a matter of running each one at the same target. Recorded in the Voice Recorder, trimmed in the Audio Cutter, normalized here: that is the order I use.
Good to know
- One loud moment decides the whole file. A cough, a door, a knuckle on the desk: whichever sample is biggest gets lifted to your target, and the speech underneath rises by that same modest amount. Cut the spike out in the Audio Cutter and run it again to see how much difference it makes.
- Peak is a different quantity from loudness. Two files normalized to 89 can still feel unequal, because ears average energy over time while a peak meter reports one instant. Streaming platforms measure LUFS, and this tool has nothing to say about landing on -14.
- Dynamics come out exactly as they went in. There is no compressor and no limiter here, so the gap between the quietest passage and the loudest one is the same gap it always was.
- Distortion already baked into a recording stays baked in. Lifting the level carries the damage up with everything else.
- Every export is written at 16-bit, FLAC included, and a lossy source picks up one more encode on the way out. That is fine for a voice memo. It is the wrong place to prepare audio for a mastering engineer.
Frequently asked questions
How much louder does this actually make a quiet recording?
As much as the headroom allows, and the sum is easy to do in advance. The gain is your target divided by the current peak. A take whose loudest sample sits at 0.2, which is roughly -14 dBFS, normalized to 89 gets multiplied by about 4.5, a lift of nearly 13 dB. Room hiss and breath noise come up by that same 13 dB, and that is the part people forget.
Should I use 100, or leave a bit of headroom?
100 is fine for something you are about to listen to once and delete. Anything with another step after it wants 89. A file sitting at exactly full scale has no room left for the next encoder, and lossy formats can reconstruct a sample slightly above where it started, which is where crackle on playback tends to come from.
Why is my file still quieter than a song on Spotify?
Commercial masters are compressed and limited before release, which drags their average level up close to the peak and holds it there. Peak normalizing moves the single loudest sample to your ceiling and leaves the average exactly where it was. Same peak, wildly different perceived loudness. Closing that gap takes a limiter, which this page deliberately does without.
I ran it and the result sounds identical. What happened?
Check the peak of what you loaded. A recording already touching full scale gets a gain of 1, which is arithmetic for leaving it alone, and a file that is silent the whole way through comes back untouched because there is no peak to scale. Running the same file through twice does nothing for the same reason.
Missing something in Audio Normalizer? Suggest a feature →