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Trim Audio Online Free

Trim audio to the exact millisecond.

Drag the handles on the waveform or type the times, loop the selection until it sounds right, then save it. Decoded, cut and encoded in your browser.

audio_fileOpen audiomusic_noteSample track

openssaves

.mp3.wav.ogg.flac.aac.m4a.webm.opus.mp4

content_cut

.wav16-bit PCM.m4aAAC 128k.webmOpus 128k.oggnot here

1 mscut precision

Looppreview the cut

On deviceno upload

graphic_eqsample-track.wav (generated) · 0:48.00 · 44.1 kHz · stereoclip0:15.000

0:18.24

play_arrowrepeatLoop

start−10−1+1+10

end−10−1+1+10

Drag across the waveform, type a time (1:23.456 or 83.456), or nudge by a millisecond. Enter applies a typed time. Fades are drawn on the waveform as you set them.

FadesFade in0.30 sFade out3.00 s

LinearEqual powerExponentialS-curve

Output format

.wavlossless, 44.1 kHz 16-bit.m4aAAC 128 kbps.webmOpus 128 kbps.oggnot in this browser

length0:15.00

size ≈2.65 MB

true peak-0.30 dBTP

RMS level-15.4 dBFS

samples661,500

from0:18.24

warningTrue peak is above −1 dBTP; normalise for streaming and lossy formats.downloadCut & download .wav

Quick clips from the start point:phone_iphoneRingtone · 30 snotificationsNotification · 3 sphoto_cameraStory · 15 spodcastsIntro · 20 smusic_noteChorus · 45 s

~/how-long-can-it-be

Length limits, checked against your clip.

Your current selection is 0:15.00. Limits are typical as of 2026 and change without notice.

phone_iphoneiPhone ringtone

check_circle40 s max

Ringtones over 40 s won’t sync. Needs AAC in an .m4r file.

smartphoneAndroid ringtone

check_circle1 min max

Any length works; 20-30 s loops nicely.

notificationsNotification sound

cancel5 s max

Short and distinct: under 5 s.

photo_cameraInstagram Story music

cancel15 s max

Each Story segment plays 15 s.

podcastsPodcast intro

check_circle30 s max

Most shows keep intros under 30 s.

chatWhatsApp voice clip

check_circle15 min max

Long clips are fine; smaller files send faster.

~/shape-of-a-fade

Same length, four different fades.

Each one fades the last 3.0 s of your clip. The dashed line is where it starts to sound half as loud. Click a card to use it.

Linear

52%in, half as loud

Hangs on, then cuts off

Equal power

68%in, half as loud

Smooth, the usual pick for music

Exponential

13%in, half as loud

A steady, even fade

S-curve

52%in, half as loud

Gentle at both ends

~/peaks-between-samples

Safe on screen can still clip after export.

Dots are the samples in your file. The line is the wave a player rebuilds, and it can rise above the dots. That hidden peak is the true peak; keep it under −1 dB.

Your selectionwarningToo hot

Loudest sample-0.3 dBTrue peak-0.3 dB

Turn it down 0.7 dB to be safe. Normalise does it for you.

Worst-case test tonewarningWill clip

Loudest sample-0.2 dBTrue peak2.9 dB

The samples look safe, but the real wave is 3.1 dB higher.

~/cutting-without-re-encoding

A quick cut snaps to the nearest frame.

Compressed audio is stored in small blocks called frames. Cutting without re-encoding keeps whole frames, so your start and end move a little. This page re-encodes, so it cuts exactly where you chose.

How far your selection would movequick cut

formatframestartend

MP326.1 ms−6.5 ms+13.9 ms

AAC-LC23.2 ms−12.3 ms+11.0 ms

HE-AAC46.4 ms−35.6 ms+11.0 ms

Opus20.0 msexactexact

Vorbis46.4 ms−35.6 ms+11.0 ms

FLAC92.9 ms−35.6 ms+11.0 ms

The same cut, three ways

Quick copysnaps to frames

ffmpeg -ss 18.240 -to 33.240 -i in.mp3 -c copy out.mp3

Re-encodeexact · small

ffmpeg -i in.mp3 -ss 18.240 -to 33.240 -c:a libopus -b:a 128k out.opus

Uncompressedexact · large

ffmpeg -i in.mp3 -ss 18.240 -to 33.240 -c:a pcm_s16le out.wav

~/which-format

Pick the format by where it’s going.

Sizes are for your current 0:15.00 selection. The last column is your browser, tested live.

FormatQualityYour clipBest forThis browser

.wavperfect · lossless≈ 2.65 MBEditing later, video projects, archiving the cut.can export

.flacperfect · lossless≈ 1.46 MBFull quality at roughly half the size of WAV (music; varies).decode only

.m4avery good · lossy≈ 240 KBiPhone ringtones (as .m4r) and Apple devices.can export

.mp3good · lossy≈ 240 KBAnything that must play everywhere, including old car stereos.decode only

.webmvery good · lossy≈ 240 KBWeb pages and browsers; Opus is the best quality per bit here.can export

.oggvery good · lossy≈ 240 KBGames, Android notifications and open-source players.can’t encode

Which one for this clip?

edit_noteEditing it furtherLossless 16-bit PCM imports cleanly into any editor, without a second round of lossy compression on top of the source..wavselected

sendSharing in chat or on a pageOpus at 128 kbps is about a tenth of the WAV size and plays in every current browser..webmuse this · 0.24 MB

devicesNot sure where it will openWAV plays in virtually every audio app and needs no encoder in the browser, so it always works..wavselected

phone_iphoneiPhone, older Safari, iTunesAAC in M4A is the native Apple format. Where the browser cannot encode it, use WAV..m4ause this · 0.24 MB

Timeline

1877Edison’s phonograph; cutting meant physically splicing later media.

1948Magnetic tape editing with razor blades and splicing blocks begins in radio.

1991MP3 is finalised as MPEG-1 Layer III, with 1,152-sample frames.

1997AAC is standardised in MPEG-2 Part 7; 1,024-sample frames.

2000Vorbis 1.0 development is underway; Ogg Vorbis 1.0 ships in 2002.

2004mp3DirectCut and similar tools popularise frame-accurate cutting without re-encoding.

2006ITU-R BS.1770 defines loudness and true-peak measurement with 4× oversampling.

2010EBU R 128 recommends −1 dBTP maximum for broadcast.

2012Opus becomes RFC 6716, with 2.5-60 ms frames.

2015Web Audio lets browsers decode, cut and play audio without plugins.

Sources

ITU-R BS.1770Loudness and true-peak measurement; 4× oversampling.north_east EBU R 128Loudness normalisation and the −1 dBTP limit.north_east ISO/IEC 11172-3 (MP3)Layer III frames of 1,152 samples and the bit reservoir.north_east IETF RFC 6716: OpusFrame sizes from 2.5 to 60 ms.north_east W3C: Web Audio APIdecodeAudioData, OfflineAudioContext, AudioBufferSourceNode.north_east W3C: MediaStream RecordingMediaRecorder formats used for compressed export.north_east FFmpeg documentation: -ss and -c copySeeking and stream copy behaviour.north_east