Trim audio without re-encoding it

Drop an audio file here

or click to choose — MP3, M4A, WAV, FLAC, OGG, Opus, AAC

Choose the part to keep on the waveform or type the times, then cut. The audio is copied frame by frame, not re-encoded, so it sounds exactly like the original. A lossless cut can only fall between the small frames that MP3 and other compressed formats are made of; this tool shows where your cut actually landed, to the millisecond, and adds the short lead-in an MP3 needs so the first note isn't garbled. For a cut on the exact sample, save as WAV.

How to trim an audio file

  1. Drop an MP3, M4A, WAV, FLAC or Opus file on the box above.
  2. Drag across the waveform to select the part to keep, or type the start and end times. Play it to check; Use playhead sets a time from where you paused.
  3. Press Trim audio, listen to the result and download it. The box under the player shows where the cut landed and how many frames were copied unchanged.

Why a lossless cut can't fall on any millisecond

Compressed audio is stored in small blocks called frames, and each frame is decoded as a whole. Cutting without re-encoding means keeping whole frames, so the cut can only land on a frame boundary. How far apart those are depends on the format:

FormatFrame lengthWhat the cut does here
MP31,152 samples — 26.1 ms at 44.1 kHzStarts at the frame boundary, plus 2 frames of lead-in
M4A (AAC)1,024 samples — 23.2 ms at 44.1 kHzExact: the file tells players where to start
Opus, Oggusually 20 msStarts at the packet boundary, plus 80 ms of lead-in
FLACusually 4,096 samples — 93 msStarts at the frame boundary; FLAC frames stand alone
WAVno framesExact to the sample

A few milliseconds either way is rarely audible as a timing difference. It does mean that a trimmer which cuts an MP3 on any millisecond you like has re-encoded it. This tool keeps the original frames and tells you, to the millisecond, where the start and end actually are.

The MP3 catch: the first frame after a cut

MP3 encoders save space by letting a frame keep part of its data in the frame before it (the "bit reservoir"). Cut exactly on a frame boundary and the first frame of the new file is missing that borrowed data. We measured it on a test MP3: with a plain frame-boundary cut, the first 35 milliseconds after the cut point decoded wrongly. Keeping two frames before the cut (52 ms) fixed it completely: from the first sample of the chosen start, the trimmed file decoded identically to the original.

So an MP3 trimmed here begins 52 to 78 milliseconds before your start time. That lead-in is part of the recording you chose to cut from, not added noise, and the result box says how long it is.

What we measured

Each format was cut from 5.300 to 12.700 seconds of the same 20-second test recording. The trimmed file was decoded by the browser and lined up against the original, sample by sample:

FileTrimmed file starts atFrames copied unchangedSound from 5.300 s on
MP3, 128 kbps5.224 s287 of 287Identical
M4A, AAC5.300 s320 of 320Identical
FLAC5.294 s80 of 80Identical
WAV5.300 s—Identical
Opus (libopus)5.220 s375 of 375Identical after 0.3 s

Opus frames are copied unchanged too, but its decoder carries a little state from one packet to the next and needs a moment to settle. In the first 0.3 seconds after the cut the sound differed slightly from the original, by at most 6% of full scale, which is why Opus gets 80 ms of lead-in.

When to choose the WAV option

Choose Cut to the exact sample when the start has to be exact: a loop that must repeat without a gap, a sound effect lined up with a video frame, or a sample for music software. The part you keep is decoded and saved as 16-bit WAV, starting and ending exactly where you said; in our tests it started at 5.300 s from MP3, M4A and Opus alike. WAV is uncompressed, about 10 times the size of a 128 kbps MP3. To cut a video rather than its sound, use Trim Video; to take the sound out of a video first, use Extract Audio from Video.

The audio never leaves your device: it is read, cut and saved in your browser.

Frequently asked questions

Does trimming lower the quality?

Not with the default Keep the format option. The compressed audio frames are copied into the new file exactly as they are, and the result box counts them: in our tests every frame of the trimmed MP3, M4A, FLAC and Opus files was a byte-for-byte copy of the original. Re-encoding, which many trimmers do, compresses the sound a second time and loses a little more each time.

Why does my trimmed MP3 start slightly before the time I chose?

MP3 audio is stored in frames of 1,152 samples, 26.1 milliseconds at 44.1 kHz, and a cut without re-encoding can only fall between frames. The tool starts at the frame that contains your start time and keeps two frames before it, because an MP3 frame often borrows data from the one before. Without them, the first 35 milliseconds after the cut decoded wrongly in our test. The result box shows the exact times.

Can I cut on the exact millisecond?

Yes: choose Cut to the exact sample. The chosen part is decoded and saved as an uncompressed WAV, so it starts and ends exactly where you said, to the sample. The catch is size: CD-quality WAV is about 10 times larger than a 128 kbps MP3. M4A and WAV files are cut exactly even in the default mode.

Which audio formats can I trim?

MP3, M4A and other AAC files, WAV, FLAC, and Opus or Ogg files such as WhatsApp and Telegram voice messages. The trimmed file keeps the original format. Formats this tool can only read, not copy, can still be cut with the WAV option, as long as your browser can play them.

How do I choose the part to keep?

Drag across the waveform to select a part, or drag the edges of the selection. You can also play the file, pause where you want to cut and press Use playhead, or type the time as seconds (65.5) or minutes and seconds (1:05.5).

Is my audio uploaded?

No. The file is read and cut inside your browser, so voice notes, interviews and recordings stay on your device. Nothing is sent to a server.