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Audio Trimmer & Converter

Cut audio on a waveform, add fade in and fade out, normalise the peak, change speed or mix to mono, then export WAV (16, 24 or 32-bit float) or Opus. Preview the result before saving.

Overview

Trims and edits audio on a waveform in your browser. Select a region by dragging on the waveform or with sliders, add fade in and fade out, normalise the peak level, change speed, mix to mono, preview the result and export WAV at 16, 24 or 32-bit float, or Opus where supported. Files are decoded locally and never uploaded.

How It Works

Drop an audio file, or a video to use its soundtrack. A waveform appears with the selected region highlighted between two red markers. Drag on the waveform to move the nearest marker, or use the Start and End sliders for keyboard control. Set fade in and fade out times, a speed from 0.5x to 2x, normalise the peak to a level such as -1 dBFS, and tick Mono if wanted. Preview result plays exactly what will be exported, and Export saves the file with -edit added to its name.

Step-by-Step Usage Guide

  1. Drop an audio file such as MP3, M4A, WAV, OGG or FLAC.
  2. Drag on the waveform or use the sliders to select the part to keep.
  3. Add fades, change speed, normalise or switch to mono as needed.
  4. Click Preview result to listen to the edit.
  5. Choose WAV or Opus and click Export, then download.

Technical Specifications & Standards

The file is decoded with the Web Audio API into floating-point samples per channel, and every edit is plain arithmetic on those samples, so the output is exact to the sample. The waveform is drawn from 800 buckets of minimum and maximum values. The region is cut by sample index, mono mixes channels by averaging, and the speed change resamples with linear interpolation. Like a tape played faster, that also raises the pitch; it is not a time-stretch. Fades are linear ramps over the chosen seconds, and normalisation scales the whole clip so its loudest sample reaches the target level, which changes peak level, not perceived loudness. WAV is written by this tool as 16 or 24-bit integer PCM, or 32-bit IEEE float with the fact chunk the format requires, which keeps samples without rounding. Opus is recorded through MediaRecorder at 160 kbps in real time. The peak level panel shows the source's loudest sample in dBFS, so you can spot clipping before you normalise.

Targeted Use Cases

  • Cutting a ringtone or notification sound from a song.
  • Trimming silence from the start and end of a podcast or voice recording.
  • Adding a fade out to a music clip for a video edit.
  • Raising the level of a quiet recording before sharing it.
  • Converting an M4A or OGG file to WAV for software that needs it.

Notes & Gotchas

  • Normalise to -1 dBFS rather than 0 to leave headroom for later lossy encoding.
  • Use 24-bit or 32-bit float WAV if you will keep editing in another program.
  • Add a fade of at least 0.05 seconds at hard cuts to avoid clicks.
  • Remember that speed changes also shift pitch, which suits effects more than music.

Frequently Asked Questions

Can it export MP3?

No. Browsers include no MP3 encoder and the site does not upload audio to a server. Export WAV, which any audio app can convert to MP3, or Opus, which is smaller than MP3 at the same quality.

Does changing the speed change the pitch?

Yes. The tool resamples the audio, like playing a tape faster or slower, so 1.25x is also higher in pitch. Keeping pitch while changing tempo needs a time-stretch algorithm, which this tool does not include.

What does normalise do?

It scales the whole clip so the loudest sample reaches the level you pick, such as -1 dBFS. Quiet recordings get louder without clipping. It sets peak level, not the loudness standards streaming services use.

Is there a file size limit?

Files over 1 GB are refused, because the whole decoded clip must fit in memory. An hour of stereo audio decodes to about 1.3 GB of samples, so very long recordings may be slow on phones.

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