Audio Trimmer

About Audio Trimmer

This online audio trimmer allows you to cut and trim audio files directly in your browser. Simply select an audio file, choose the start and end time using the timeline or manual input, and trim your audio. All processing happens locally for complete privacy. See also our Audio Joiner and Audio Converter.

How do I trim or cut an MP3 file to keep only a specific section?

Load your audio file, then set the part you want to keep using the dual-handle timeline slider, the 'Set as Start' and 'Set as End' buttons (which capture the current playhead position), or by typing precise timecodes into the Start Time and End Time fields. The preview player plays only the selected range so you can verify it before exporting. Click 'Trim Audio' to save just that section as a new file. The tool runs entirely in your browser via FFmpeg.wasm, so your files never upload to a server. Trimming is lossless when you keep the source codec and use Copy quality (MP3 in, MP3 out).

What audio formats can I trim with this tool?

You can trim MP3, WAV, FLAC, OGG Vorbis, AAC, M4A, Opus, WebM audio, AIFF, and the audio track of common video files (MP4, MKV, MOV) — essentially any container FFmpeg.wasm can decode. By default the output keeps the source format, but you can re-encode to MP3, WAV, OGG, or AAC during the trim if you need to change codecs. WAV and FLAC are sample-addressable, so any cut point works; lossy formats like MP3 and AAC are frame-based, which matters when you choose between Fast and Accurate cut precision.

Will trimming reduce the audio quality of my file?

Not if you use Copy quality with Fast precision — that mode copies the encoded frames straight through without decoding or re-encoding, so quality is bit-identical to the source (truly lossless). Quality only changes when the tool must re-encode: that happens when you pick a bitrate (High/Medium/Low), apply a fade, convert to a different format, or choose Accurate cut precision. Re-encoding adds one generation of compression, usually inaudible at 192 kbps or higher. If you plan further edits, trim to a WAV intermediate first and export to MP3 only at the very end.

How precise are the cut points — can I trim to the millisecond?

Yes. The Start Time and End Time fields accept HH:MM:SS.mmm, and the tool preserves the fractional seconds end to end (no rounding to whole seconds). The 'Set as Start/End' buttons also capture the playhead position with millisecond precision. How that timecode is honored depends on the Cut Precision setting: Fast snaps the cut to the nearest frame boundary (about 26 ms for MP3) while copying frames losslessly; Accurate decodes and re-encodes so the cut lands exactly on the requested millisecond — sample-precise, at the cost of one re-encode generation. For ringtones, dialogue editing, or syncing audio to video, choose Accurate.

Can I fade in and fade out at the cut points to avoid clicks?

Yes. Cutting in the middle of a loud passage creates a sudden amplitude jump that produces an audible click or pop. Use the Fade In and Fade Out sliders to ramp the amplitude smoothly at each boundary (FFmpeg's afade filter). Quick presets are provided: 50 ms is a click-free micro-fade that is sonically invisible, 0.5 s is a soft fade, 2 s is a smooth fade, and 5 s is a cinematic fade. Speech rarely needs more than the 50 ms preset; complex stereo music usually benefits from a short fade on both ends. Enabling any fade automatically switches Copy mode to re-encode, since filters cannot be applied while stream-copying.

What is the difference between Fast and Accurate cut precision?

Fast precision performs an output seek (it places -ss before the input), which lets FFmpeg copy whole encoded frames without decoding. It is the fastest option and stays lossless with Copy quality, but the cut snaps to the nearest frame boundary — up to about 26 ms for MP3 (1152 samples per frame at 44.1 kHz) or 21 ms for AAC. Accurate precision performs an input seek (-ss after the input), forcing FFmpeg to decode to the exact requested sample before cutting; it then re-encodes, so the timecode you typed in HH:MM:SS.mmm is honored precisely. Use Fast for podcasts, lectures, and songs where frame snapping is inaudible; use Accurate for tight, sample-precise cuts.

Is my audio file private, and is there a file size limit?

Everything runs locally in your browser through FFmpeg.wasm (WebAssembly) — your file is never uploaded to any server, so it stays completely private on your device. The current size limit is 50 MB, which keeps in-browser processing fast and responsive. FFmpeg.wasm is about 30 MB and loads once per session; after that, trims with Copy quality and Fast precision complete in seconds.