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Audio, Video & Voice · Free tool

Audio Trimmer

Trim or cut any audio file in your browser. Pick start and end, preview, download the trimmed clip. No upload.

Updated June 2026
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What it does

Drop an audio file (MP3, WAV, M4A, OGG, FLAC, or anything else your browser can decode), see the waveform, drag the start and end handles to mark the section you want to keep, and export a trimmed WAV. Useful for shaving intros and outros off podcast clips before sharing, cutting a 30-second sample out of a song for a video soundtrack, removing dead silence at the start of a voice note, extracting a quote from a long interview recording, or preparing audio fixtures for a podcast editor (each clean intro / sting / sound effect as its own file).

The waveform display lets you see exactly where speech, music, or silence sit in the file — drag the handles right up against a word boundary, click to play just the selection, and verify before exporting. Output is 16-bit PCM WAV at the source sample rate, which is universally supported (every audio editor, every podcast platform, every video editor accepts WAV input).

Trim is sample-accurate: the cut points are exact to the millisecond level (versus codec-block alignment in formats like MP3 where fades can creep in at frame boundaries). The whole pipeline runs locally via the Web Audio API — no upload, no server, file stays in browser memory.

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<iframe src="https://freetoolarena.com/embed/audio-trimmer" width="100%" height="720" frameborder="0" loading="lazy" title="Audio Trimmer" style="border:1px solid #e2e8f0;border-radius:12px;max-width:720px;"></iframe>
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How to use it

  1. Drop your audio file (drag-drop or click to browse). The waveform renders below — typically takes 1-2 seconds for a 5-minute file.
  2. Drag the left handle to set the trim start, right handle to set the trim end. Click anywhere on the waveform to scrub the playback head.
  3. Click Play to hear just the selected range. Adjust handles by single pixels (= a few milliseconds) for tight trims.
  4. Click Export. The selection is sliced from the underlying AudioBuffer and re-encoded as 16-bit PCM WAV.
  5. Click Download to save the WAV file. To convert to MP3 afterward, use the audio-format-converter tool or any audio editor.

When to use this tool

  • Cutting a clean intro or outro off a podcast clip before sharing.
  • Extracting a 30-second song sample for use as background music in a short video.
  • Removing dead silence at the start or end of a voice memo.
  • Splitting a long interview into individual question/answer clips for transcription or quoting.

When not to use it

  • Multi-cut editing (remove the middle, keep start + end) — this is single-range trim. Use Audacity or a DAW for multi-cut.
  • Adding fades, EQ, compression, or other effects — that's editing. Use Audacity, Reaper, GarageBand, or a real DAW.
  • Long audio files (>30 minutes) — the waveform renders slowly and browser memory becomes a bottleneck around an hour.
  • Lossless preservation of MP3 source — output is WAV, which is uncompressed; the source is decompressed once. There's no MP3-to-MP3 lossless trim in the browser (use ffmpeg locally with `-c copy` for that).

Common use cases

  • Onboarding a colleague who needs the same calculation/conversion
  • Verifying a number or output before passing it on
  • Quick use during a typical workday
  • Pre-decision sanity-check on inputs and outputs

Frequently asked questions

Why is the output WAV and not the original format?
Browser audio APIs decode any input to PCM samples internally. The simplest reliable export from PCM is WAV (no encoder needed; it's just header + raw samples). To re-encode as MP3 / AAC / OGG you'd need an in-browser encoder, which adds significant complexity and quality tradeoffs. WAV → MP3 conversion afterward is fast and lossless within the WAV's bit depth.
How accurate is the trim point?
Sample-accurate — the cut is exact to the audio sample level (typically 1/44,100th of a second, or ~22 microseconds, for CD-quality input). That's much tighter than the keyframe-based trims in lossy formats like MP3.
Will the output preserve stereo?
Yes. Both channels are decoded and trimmed identically. The output WAV is stereo if the input was stereo, mono if it was mono.
Why does my MP3's reported duration mismatch what I see in the waveform?
MP3 files often report a slightly different duration than they actually contain due to encoder padding (silence at the start or end added during encoding). The browser shows you the exact decoded length, which is what gets trimmed. The handles use the actual audio data, not the reported metadata.
Is my audio uploaded anywhere?
No. The file is decoded into a Web Audio AudioBuffer in your browser, trimmed in memory, and re-encoded as WAV — entirely client-side. Open DevTools → Network during processing and you'll see zero outbound requests.
What's the maximum file size?
Limited by your browser's memory (~250-500 MB practical for a typical laptop). A 60-minute MP3 is ~50 MB compressed; the decoded WAV in memory is ~600 MB at CD quality. Past ~30 minutes input, things get sluggish; past an hour, you may run out of memory.

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