AudioCutlet

Cut, fade and export audio in your browser.

Open an MP3, WAV, M4A, OGG or FLAC, edit it on the waveform below and download a WAV or MP3. Nothing is uploaded — the file stays on your device.

Drop an audio file here

or
00:00.000 / 00:00.000

Operations

Export

Range
Format
Bit depth
.wav

WAV is uncompressed and exact — the right choice if you will edit again.

Everything happens in this tab. Your audio is never uploaded — you can go offline once the page has loaded.

What the editor does

Nothing is uploaded

The file is decoded by your browser's own audio engine and every edit and export runs in this tab. There is no server, no account and no upload — the page works offline once it has loaded.

Cut on the waveform

Drag a selection, type exact in and out points to the millisecond, then trim, cut or split at the playhead. Zoom with the wheel or a pinch down to individual samples. Every step can be undone.

Level, fade and clean up

Fade in and out by a chosen number of milliseconds, apply gain, normalize peaks to −1 dBFS or loudness to an RMS target, and remove pauses below a threshold — with the matches highlighted before you commit.

Export WAV or MP3

Download the whole file or just the selection as 16- or 24-bit WAV, or as MP3 at 128, 192 or 320 kbps. Split points turn the file into segments you can download one by one or as a single ZIP.

About AudioCutlet

AudioCutlet is an online audio cutter that does its work without a server. Open an MP3, WAV, M4A, AAC, OGG or FLAC file and your browser decodes it with the Web Audio API; the waveform you see is drawn from the real samples, not a preview generated somewhere else. From there you can trim a song down to a chorus, cut a cough out of a recording, split a long session into named parts, add a fade in and fade out, raise or lower the level, normalize the loudness of several clips to the same target, or strip the dead air from a voice memo. Each operation writes a new version onto an undo stack, so the original decode is always one click away.

Export is deliberately simple. WAV is written by the editor itself as 16- or 24-bit PCM, so what you download is exactly what you heard. MP3 is encoded in the page with the LAME encoder compiled to JavaScript, at 128, 192 or 320 kbps, with a progress bar because a long track takes a few seconds. Because nothing leaves your device, the practical limit is your computer's memory rather than an upload quota: a stereo song takes roughly ten megabytes of memory per minute once decoded, which is comfortable for full albums on a laptop and for single tracks on a phone. The keyboard does most of the work if you let it — press ? to see the shortcuts.

Guides

How cutting, loudness and ringtones really work — written to be read once and remembered.

All guides

Questions people ask

What people usually want to know before they open a file — honest answers, including the things the editor does not do.

Is my audio uploaded anywhere?

No. The file is read by your browser and decoded with the Web Audio API on your own device; every edit is arithmetic on the decoded samples in the tab's memory, and the WAV writer and MP3 encoder run there too. The site has no server-side component and makes no network requests after the page has loaded — you can switch to airplane mode and keep working.

Which formats can I open?

Anything your browser can decode: MP3, WAV, AIFF and M4A/AAC everywhere; OGG Vorbis and Opus in Chrome, Edge and Firefox; FLAC in Chrome, Edge, Firefox and Safari 17 or newer. The audio track of an MP4 or WebM video also opens. Files protected by DRM — offline downloads from streaming apps — cannot be decoded by any browser, and the editor says so instead of guessing.

Does exporting to MP3 lose quality?

One generation, yes. An MP3 has to be decoded to edit it, and the export encodes the result again, so a little more detail is discarded. At 192 kbps the difference is inaudible to most people for a single pass; at 320 kbps it is smaller still. If you plan to edit the file again later, export WAV — it is exact — and make the MP3 from the WAV at the very end.

How long a file can I edit?

The limit is memory, not a quota. Decoded stereo takes about 10 MB per minute (44.1 kHz, 32-bit float), and the undo stack keeps up to twelve versions, so a busy edit of a one-hour recording can use a few gigabytes. A laptop handles an hour comfortably; a phone is happier with ten to fifteen minutes. Files over 250 MB are refused up front, because a 250 MB MP3 would decode to several gigabytes.

Can I export M4A, AAC, OGG or FLAC?

No — the editor writes WAV (16- or 24-bit) and MP3 (128, 192 or 320 kbps). Browsers ship decoders for the other formats but no encoders, and bundling one would make the page several megabytes heavier. If you need an M4A (for an iPhone ringtone, say), export WAV and convert it with the Music app, iTunes or ffmpeg.

What keyboard shortcuts are there?

Space plays and pauses; I and O set the in and out points at the playhead; S splits; Delete cuts the selection out; [ and ] apply the fade in and fade out with the current length; + and − zoom, 0 fits the whole file; the arrow keys nudge the playhead by a second (a tenth with Shift); L toggles looping; ⌘Z or Ctrl+Z undoes and Shift adds redo. Press ? on any page to see the full sheet.

Does it work on a phone?

Yes. Drag on the waveform to select, tap to move the playhead, pinch to zoom and swipe the thin bar under it to scroll; the in and out fields accept typed times when a finger is not precise enough. The file picker opens the Files app on iOS and the system picker on Android, and exported files land in Downloads. Long files are limited by the phone's memory, so keep them to a quarter of an hour or so.

How accurate is Remove silence?

It measures RMS level in 10 ms windows and treats a stretch as a pause when every window stays under the threshold (−40 dBFS by default) for at least the minimum duration (500 ms by default). Matching pauses are highlighted on the waveform before you apply anything, and 80 ms is kept at each end so words do not run into each other. For a noisy room recording raise the threshold to around −30 dB; for a clean studio file you can go lower. It cannot tell a breath from background noise — listen once after applying, and undo if a pause was musically important.

Why did loudness normalization say it was capped?

Because reaching the target would have pushed the loudest samples past 0 dBFS, which clips. The editor raises the gain as far as it can while keeping peaks under −0.1 dBFS and tells you how much it applied. Getting a very dynamic clip up to −14 dB RMS without clipping needs compression or limiting, which this editor deliberately does not do; peak normalization to −1 dBFS is the safe alternative.