Normalize audio

Bring files to the same loudness, −14 LUFS or another, without clipping, on your device. Free, with no account or uploads.

Drop audio files to normalize

Each file is measured and brought to the same loudness (−14 LUFS for Spotify, or another target), with peaks kept under −1 dBTP. Many at once, on your device. You can also paste with ⌘V or Ctrl+V.

Choose files

MP3
M4A
Opus
FLAC
WAV
AIFF
MP4
MOV
WebM

Stays on your device · up to 50 files at once

This page is the Audio editor, opened for normalizing loudness.

What you can do

Trim, fade and change the volumeTo the millisecond, on a waveform you can zoom; every edit can be undone, and your file is never changed.
Every common formatMP3, M4A (AAC), Opus, FLAC, WAV and iPhone ringtones, from those and from AIFF, Ogg and the sound of videos.
Loudness to a standardMeasured as ITU-R BS.1770 does, normalized to −14 LUFS or another target, with a true-peak limiter so nothing clips.
Sound out of videoKept exactly as it is, without being encoded again, whenever the format allows; or converted to the format you choose.
Checked before you downloadEvery file written is opened again and listened to: its length, rate and channels are held against what you asked.
Many files at onceTags and cover art kept, gapless MP3s, a size limit if you need one, and everything in one ZIP.
Nothing is uploadedYour files are read and written in this browser. No account, no limits.

A little help along the way

Follow the steps below. Every option is explained underneath.

Add your filesDrop them, choose them or paste them: MP3, M4A, Opus, FLAC, WAV, AIFF or videos. Up to 50 at once.
Choose the loudness−14 LUFS for Spotify, −16 for Apple Podcasts, −23 for broadcast, or your own; and the peak ceiling.
Let each file be measuredEach is measured as it will be written and made louder or quieter to reach the target; the limiter holds its peaks.
Check and downloadEach file written is measured again; the page says what it reached. Download one, or all as a ZIP.

Every option explained

Normalize loudness: Each file is measured as it will be written, after its edits, at its new rate and channels, as ITU-R BS.1770 measures loudness (K-weighted, gated), and made louder or quieter to reach the target: −14 LUFS for Spotify, streaming; −16 LUFS for Apple Podcasts; −23 LUFS for Broadcast (EBU R 128); −24 LUFS for US TV (ATSC A/85), or one of your own. Files normalized together come out equally loud. A file is made at most 30 dB louder.
The limiter and the ceiling: When a file is made louder, its loudest moments could go over full scale and clip. The limiter looks 5 ms ahead and lowers the gain smoothly just before them, letting go over 100 ms, so peaks stay under the ceiling, between samples too (true peak). −1 dBTP leaves room for MP3 and AAC encoders, which raise peaks a little. If the ceiling keeps a file from its target, the page says how loud it reached.
Normalize peaks: Makes each file louder or quieter so its highest true peak reaches the level chosen, with no limiter. Files keep their own loudness, so they will not sound equally loud.
Make louder: Makes each file louder by up to 30 dB, with the limiter. 6 dB doubles the height of the waveform; 10 dB sounds about twice as loud.
Loudness, range and peaks: The editor shows each file's loudness in LUFS, its loudness range in LU (how much it varies, as EBU Tech 3342 measures it) and its highest true peak in dBTP, for the whole file and for the part chosen, with its short-term loudness drawn under the waveform. The check measures the file written again.
Equaliser: Raise or lower the bass (a shelf under 120 Hz), the mid (around 1 kHz) and the treble (a shelf over 8 kHz) by up to 12 dB, and cut rumble under 80 or 120 Hz. Presets: Flat (The sound's own balance, unchanged); Voice clarity (Rumble under 80 Hz cut, a little less boom, a little more presence and air); More bass (6 dB more under 120 Hz); Cut rumble (Everything under 80 Hz cut: wind, handling noise, traffic); Telephone (Only 300 Hz to 3.4 kHz, as a phone line carries). These presets are KeepSimp's own choices.
Compressor: Turns the loudest moments down so loud and quiet parts sit closer together. Gentle: Evens out the loudest moments a little: 2:1 above −18 dBFS. Voice: Keeps speech at one level, near and far from the microphone: 3:1 above −24 dBFS, quick to act. Music: Holds a mix together, letting drums through: 2.5:1 above −16 dBFS, slower to act. A compressed sound is quieter, so choose Normalize loudness to bring it back up. When the equaliser or compressor is on and the loudness is left as it is, a limiter keeps peaks under −1 dBTP.
Same as the file: Keeps the file's own sound as it is, without decoding and encoding it again: AAC goes into an M4A, MP3 stays MP3, Opus and Vorbis go into Ogg, FLAC stays FLAC and plain samples become a WAV. Nothing is lost and it takes a moment. When the rate, the channels or a size limit you chose mean the sound must change, it is encoded again in its own kind; sound that cannot be kept (such as AC-3 from a film) becomes AAC, and the page says so.
MP3: Plays everywhere. Written with LAME at a constant bitrate, with the header that tells players the encoder's delay and padding, so the file is exactly as long as the sound. Below 32 kHz an MP3 holds at most 160 kbit/s.
M4A (AAC) and M4R: Smaller than MP3 for the same quality, and played by every phone and computer. Written with your browser's own AAC encoder, or one that comes with the page when it has none. The encoder's priming is measured and an edit list written, so the sound starts and ends where it should. M4R is the same for an iPhone ringtone, which can be at most 40 seconds long (an alert tone 30).
Opus: The best quality for its size, voice above all. It is always written at 48 kHz. Browsers, Android, VLC and recent Apple devices play it.
FLAC and WAV: Lossless: every sample is kept. FLAC is about half the size of WAV. Choose 16-bit (CD quality), 24-bit, or for WAV 32-bit floating point; FLAC holds at most 24 bits.
Quality: Small, Balanced or High set the bitrate for each format (MP3 96, 192 or 320 kbit/s; AAC 96, 160 or 256; Opus 48, 96 or 160), less for mono. Custom lets you pick the bitrate. These numbers are KeepSimp's own choices.
Sample rate and channels: Keep the file's own, or choose one. When a format cannot hold the rate asked, the nearest it can is used and the page says so. Lossy formats here are mono or stereo, so surround sound is mixed down to stereo for them.
Tags and cover art: Title, artist, album, year, genre and the cover picture go into the new file where its format holds them. Switch it off to write none.
File size limit: MP3, M4A and Opus are written at the highest bitrate that keeps each file under the limit, with room for the container; the page says what it chose. Lossless formats cannot be made to fit a size.
The check: Every file written is opened again and decoded from start to end. Its length, sample rate and channels are held against what was asked, an MP3's size against its bitrate, and a file that came out silent is caught. If something is not as asked, the page says what.
Files the browser cannot read: Some browsers' decoders lack a kind of sound (Safari's lacks FLAC in some places). The page then decodes it with the browser's own player, which plays more kinds, and goes on. AIFF is read by KeepSimp itself. A file nothing here can decode says so.
Privacy: Files are read and written in this browser and never uploaded. There is no account and no limit on how many you convert.

Read the step-by-step guide