Normalize the volume of an audio file (WAV)
Automatically adjust the volume of your WAV files to reach a target peak level (in dBFS). 100% local processing. Only works with WAV files in integer PCM (8/16/24/32-bit).
Drag and drop one or more .wav files here (100 MB max each).
0 dBFS is the maximum before clipping; -1 dBFS is a common safe value.
How it works
When recording several audio sequences at different times — a podcast episode with several speakers, a voice-over recorded across multiple takes, interviews done with different microphones or distances — the final volume often varies from one clip to another. The result when listening: the audience constantly has to turn the volume up or down. This tool finds the loudest sample (the peak) of each WAV file and applies a uniform gain to bring that peak exactly to the target value you set (in dBFS, 0 being the maximum possible before clipping). This lets you process several recordings before assembling them in an edit, with a consistent peak level between them.
What exactly is volume normalization?
Peak normalization adjusts an audio file's overall gain so its loudest sample reaches a precise target level (for example -1 dBFS), without changing the sound's internal dynamics: loud parts stay loud and quiet parts stay quiet, everything is simply shifted by the same factor. This is different from a compressor or limiter, which reduce the gap between loud and quiet passages by applying non-linear processing. Normalization only aligns peak levels across multiple files, which is enough in most simple assembly cases.
Does this tool work on MP3 files?
No, currently this tool only processes WAV files in integer PCM (8, 16, 24 or 32-bit). If your recording is in MP3 format, convert it to WAV first with our MP3 → WAV tool, then import the result here to normalize its volume.
Privacy: normalization is computed entirely in your browser via WebAssembly — your recordings never pass through a server.