What is Audio Trimmer?

Cut the start, end, or middle of any audio file. See the waveform, set exact start and end points, preview your selection, and export the trimmed result — all without uploading to any server.

The trimmer decodes the file locally and uses FFmpeg via WebAssembly for the cut, so nothing leaves the device. Drag the start and end handles on the waveform, type timestamps to the centisecond (mm:ss.cc), zoom in for precision, loop the selection while you fine-tune it, and add a fade in or fade out up to five seconds. Before saving you can keep the source format, convert to MP3, WAV, OGG, FLAC or M4A, normalize the volume, or use the one-click iPhone ringtone preset for a 40-second M4R. With the source format kept and both fades and Normalize off, the output is a lossless stream copy in the original format and bitrate; any of those options triggers a single one-pass re-encode.

How to use

1. Upload an audio file (MP3, WAV, OGG, or FLAC). The waveform visualization loads automatically so you can see the audio structure.
2. Drag the start and end markers on the waveform to select the portion you want to keep, or type exact timestamps for precision.
3. Preview the selection (hit Space to toggle play and pause), turn on Loop to audition the cut on repeat while dragging, and dial in a fade in or fade out for a soft edge. Want a different file type or a more even volume? Pick an output format and switch on Normalize first, or tap the ringtone preset for a 40-second M4R — then click Trim & Download.

When to use

• Cutting a 30-second clip from a long interview for a social media post.
• Removing the silent pause at the start of a recording before publishing.
• Extracting a clean intro hook from a song to use as a ringtone.

Result

A content creator needs a 30-second clip from a 5-minute interview recording. They upload the MP3, drag the markers to 1:15–1:45, preview the selection, and export the perfect soundbite.

FAQ

Does trimming re-encode the audio and lose quality?

Leave both fades at zero, keep the source format and Normalize off, and FFmpeg performs a stream copy at the cut points, so the audio data stays bit-identical to the source — only where the file starts and ends changes. Turn on a fade, switch to a different output format, or enable Normalize, and the selection is re-encoded once so the change can be baked in. That single round-trip is still indistinguishable from the original on listening tests.

How precise can the cuts be?

The timestamp input accepts down to centiseconds (1/100 of a second). On the waveform itself, dragging gives roughly pixel-precision, which is finer than a tenth of a second on most screens. Zoom in to get sample-level accuracy on short clips.

Why does the cut sometimes start slightly before or after where I marked?

Lossy formats like MP3 and AAC can only be cut at frame boundaries (about 26 ms for MP3). FFmpeg snaps to the nearest valid frame to avoid re-encoding. For exact-sample precision, convert to WAV first and trim that.

What output format do I get?

By default the trimmer keeps the same format as the input (MP3 in, MP3 out), which preserves the original quality and keeps the file small. You can also pick a different output format — MP3, WAV, OGG, FLAC or M4A — right before downloading, or use the iPhone ringtone preset to get an M4R, all without a separate converter step.

Is there a size limit for the file I can load?

The decoder holds the entire waveform in memory locally, so very long files (an hour-plus uncompressed) can exhaust phone RAM. On a desktop a podcast episode of 50–100 MB loads comfortably. Split or compress first if you hit a hang.

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