How the audio cutter works
This tool uses the Web Audio API built into your browser to decode audio files into raw sample data. Once decoded, you can select a precise time range using the start and end controls. The tool then slices the raw audio buffer at the sample level and encodes the result as a WAV file.
Working at the sample level means the cut itself is frame-accurate — the boundary lands exactly where you put it, not on the nearest frame or block boundary.
A note on sample rates
Browsers decode audio through an AudioContext, which runs at your output device's sample rate — commonly 48kHz. If your source file uses a different rate, such as a 44.1kHz MP3, it is resampled during decoding, before the tool ever sees it. This is a limitation of the Web Audio API rather than a choice we make.
In practice the difference is inaudible, but it does mean the output isn't a bit-exact copy of the source samples. The tool displays the decoded sample rate alongside the file details, and the WAV it produces uses that same rate.
Supported formats
The Web Audio API can decode most common audio formats including MP3, WAV, OGG, FLAC, AAC, and M4A. The exact format support depends on your browser — Chrome and Firefox handle the widest range, while Safari may have limited OGG support.
Why the output is WAV
Re-encoding to MP3 would mean a second generation of lossy compression on audio that has already been compressed once. WAV avoids that. The cost is file size: uncompressed stereo audio runs about 10MB per minute, so the tool shows the estimated output size before you trim and caps a single operation at 500MB.
Everything runs in your browser. Your audio file is decoded in memory, processed, and the result is generated locally. No data is sent to any server at any point.