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Audio Trimmer

Free online audio trimmer to cut and trim audio files. Select start and end time, preview in real-time. Supports MP3, WAV, OGG, AAC. No software installation required. Fast and easy audio trimming.

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Drag & drop an audio file here
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Choose an audio file to trim (MP3, WAV, OGG, AAC, M4A)

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?

Upload your audio file, then drag the start and end handles on the waveform to select the segment you want to keep. You can also type precise timestamps in the start and end fields, or click on the waveform while the audio plays to set markers in real time. Use the preview button to listen before exporting. Click Trim to save only the selected segment as a new file. The tool runs entirely in the browser via the Web Audio API and FFmpeg.wasm, so your files never upload to a server. Trimming is lossless when source and target codecs match (MP3 in, MP3 out) and the cut falls on a frame boundary.

What audio formats can I trim with this tool?

You can trim MP3, WAV, FLAC, OGG Vorbis, AAC, M4A, Opus, WebM audio, and AIFF — essentially any container the browser can decode. The output keeps the source format by default, but you can also re-encode to a different format during the trim if you need to change codecs. For lossy formats like MP3 and AAC, trimming on exact frame boundaries avoids any quality loss — the tool will snap cut points to the nearest frame edge for sample-perfect results. For WAV and FLAC, any cut point works because these formats are sample-addressable, not frame-addressable.

Will trimming reduce the audio quality of my file?

Only if the tool re-encodes during the cut. When you trim a WAV or FLAC, quality is perfectly preserved — these are sample-accurate formats. When trimming MP3 or AAC, the cleanest approach is frame-aligned cutting, which copies frames without decoding and re-encoding them; this loses zero quality but limits cut precision to about 26 ms (one MP3 frame). For sample-accurate cuts on lossy formats, the tool decodes, trims, and re-encodes — adding one generation of compression artifact, usually inaudible at 192 kbps or higher. If you plan further edits, trim a WAV intermediate first and only export to MP3 at the very end.

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

Yes. For WAV, FLAC, and any uncompressed format, you can cut at any sample (about 22.7 microseconds at 44.1 kHz). For MP3 the smallest natural unit is one frame, typically 26 ms at 44.1 kHz (1152 samples per frame); for AAC it is 21 ms (1024 samples). For sub-frame precision on lossy files, the tool re-encodes the boundary frames so the final cut is sample-accurate. Type timestamps in HH:MM:SS.mmm format for best precision, or use arrow keys to nudge the cursor one sample at a time. Zooming the waveform reveals individual samples once you are within a few milliseconds of the target.

Audio Trimmer — Free online audio trimmer to cut and trim audio files. Select start and end time, preview in real-time. Supports MP3, WA
Audio Trimmer

Can I crossfade or fade in/out at the cut points to avoid clicks?

Yes. Cutting in the middle of a non-zero waveform creates a sudden amplitude discontinuity that produces an audible click or pop. The tool offers two solutions: zero-crossing snap automatically moves the cut point to the nearest sample where the waveform crosses zero, eliminating the discontinuity, and short fades (1 to 10 ms) ramp the amplitude smoothly at the boundary even when zero crossings are scarce (e.g., in dense music). For seamless looping or joining of trimmed segments, use a 10 to 50 ms equal-power crossfade rather than a hard cut. Speech and sparse audio rarely need fades; complex stereo music almost always benefits from a 5 ms fade-in and fade-out.

How do I remove silence from the beginning and end automatically?

Enable the auto-trim silence option. The tool scans from the start and end of your file looking for the first sample whose level exceeds a threshold (default -40 dBFS), then sets the cut points just before and after that range. You can tune the threshold: -50 dBFS is conservative (catches quiet recordings), -30 dBFS is aggressive (good for noisy studio recordings with background hiss). A short padding (typically 100 to 250 ms) is preserved at each end to avoid clipping the natural attack or decay. For mid-file silence removal — between verses, between podcast segments — use the silence-trimmer or silence-detector tool, which handles multiple regions at once.

Can I trim multiple segments and join them into one file?

Yes. Add multiple selection regions to the waveform — each region becomes one kept segment, and everything outside the regions is discarded. The tool concatenates the kept segments in timeline order into a single output file. This is the classic workflow for editing out filler words from a podcast, removing applause between songs in a concert recording, or making a highlight reel from a long voice memo. Apply short crossfades between joined segments to mask the splice — without them, abrupt timbral or background-noise shifts at each boundary will be obvious. For more than 10 segments, consider exporting to a DAW for finer control.

What is the difference between sample-accurate and frame-accurate trimming for MP3?

MP3 stores audio in fixed-size frames of 1152 samples (26.1 ms at 44.1 kHz). Frame-accurate trimming copies whole frames from the source without decoding, so the cut point lands on the nearest frame boundary — fast, lossless, but with up to 26 ms of imprecision. Sample-accurate trimming decodes the source PCM, cuts at the exact requested sample, and re-encodes — bit-precise timing at the cost of one extra generation of lossy compression on the boundary frames. For long-form trimming (podcasts, lectures, songs) frame accuracy is invisible; for music production, ringtones, dialogue editing, or anything requiring tight sync to video, use sample-accurate mode. The same principle applies to AAC with a 1024-sample frame (21 ms at 48 kHz).