Trim, cut, and slice audio files with interactive waveform visualization. Drag handles to select portions, use keyboard shortcuts, zoom and pan, preview selection before export. Supports MP3, WAV, OGG, AAC.
Audio trimming is a sample-level operation: you pick a start sample and an end sample in the source recording and keep only what falls between them. A 44.1 kHz stereo file has 88,200 samples per second (two channels at 44,100 samples each), so a three-minute recording contains about 15.9 million samples in the audio stream. Trimming to a 10-second clip keeps 882,000 of those samples and discards the rest. The kept samples carry the original audio; the only place trimming can introduce a small artifact is at the boundaries, where cutting mid-waveform can produce an audible click because the signal jumps from some nonzero value to silence. The waveform display plots amplitude versus time, averaged into pixel-wide bins. Tall peaks show loud sections, flat lines show silence, and the detailed shape helps you place the start and end handles at natural breaks, the end of a spoken sentence, the tail of a musical phrase, the gap between claps. You can zoom in (Ctrl + scroll) for finer placement and type exact start and end times in seconds for sample-region precision. To avoid a boundary click, aim your cut at a quiet point or add a short fade in a separate step.
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Detect the tempo (BPM) of any audio file. Includes tap tempo feature and genre reference guide.
You recorded a 6-minute voice memo but only the 0:42–1:15 section is the usable quote.
Input
memo.m4a · keep 00:42.0 → 01:15.5
Output
memo_clip.m4a, 33.5 s, sample-accurate cut, no re-encode artefacts
Trimming on the waveform lets you set the in/out points precisely at zero-crossings, avoiding the click an arbitrary cut introduces. Processing is in-browser, so a private recording is never uploaded.
Audio trimming is a sample-level operation: you pick a start sample and an end sample in the source recording and keep only what falls between them. A 44.1 kHz stereo file has 88,200 samples per second (two channels at 44,100 samples each), so a three-minute recording contains about 15.9 million samples in the audio stream. Trimming to a 10-second clip keeps 882,000 of those samples and discards the rest. The kept samples carry the original audio; the only place trimming can introduce a small artifact is at the boundaries, where cutting mid-waveform can produce an audible click because the signal jumps from some nonzero value to silence. The waveform display plots amplitude versus time, averaged into pixel-wide bins. Tall peaks show loud sections, flat lines show silence, and the detailed shape helps you place the start and end handles at natural breaks, the end of a spoken sentence, the tail of a musical phrase, the gap between claps. You can zoom in (Ctrl + scroll) for finer placement and type exact start and end times in seconds for sample-region precision. To avoid a boundary click, aim your cut at a quiet point or add a short fade in a separate step.
Isolate the catchiest few seconds of a song and export at the right length for a phone ringtone.
Pull a highlight or quote from a long episode to share on social media.
Strip dead air from the beginning and end of a recording before archiving or sending it.
Cut loops and one-shots from longer recordings for use in a DAW or sampler.
The trimmer decodes your file in the browser with the Web Audio API so it can render the waveform and let you preview the selected region instantly. When you click Trim and Save, the actual cut is performed by FFmpeg compiled to WebAssembly: it seeks to your start time and writes out exactly the selected duration. The export format is MP3, WAV, or OGG; if the source extension is one of those it is preserved, otherwise the output defaults to MP3. The cut is always re-encoded by FFmpeg (there is no raw stream-copy path), which is why exporting to WAV is the choice to make when you want the cleanest, uncompressed result from the selected range.
Why re-encode rather than copy bytes? Most compressed formats do not allow a clean cut at an arbitrary sample, MP3 frames are 1152 samples each and can reference state from previous frames, so slicing mid-frame would corrupt the decode. Letting FFmpeg decode and re-encode the selected span produces a valid, gap-free file. For a trim where you will loop or butt the clip against another segment, place your start and end at quiet points, or export and then apply a short fade in/out, to keep the boundaries click-free; this is the practical equivalent of the zero-crossing snap that DAWs use. Including 50-100 ms of low-level audio at each boundary also gives a later fade something to taper smoothly.
No. The samples you keep are identical to the original. Quality only changes if you re-encode to a different format or lower bitrate on export.
There is no fixed limit. Large files may take a moment to render the waveform depending on your device.
Currently you select a single continuous region per trim. For multiple cuts, export each segment separately.
Audio is decoded and processed locally with the Web Audio API. Your files are never uploaded to a server.