Audio Format Converter

Convert between WAV, FLAC and MP3, and change sample rate or bit depth.

Your files stay on your device. All processing runs in your browser using WebAssembly — nothing is uploaded to a server. You can disconnect from the internet after this page loads and it will still work.

Drop an audio file here
or click to browse
WAV, MP3, FLAC, OGG, M4A — anything your browser can decode.
2. Output format
3. Settings
Every level is lossless. Higher just spends more time searching for a smaller file; what you decode is identical either way.
Adds a tiny amount of noise when reducing to 16-bit, which trades an inaudible noise floor for the loss of quantisation distortion. Right for a finished master; leave it off for an intermediate file you will process further, since it makes the conversion no longer bit-exact.

Result

Format
 
Sample rate
kHz
Size
MB
Channels
 

Your browser cannot play this format back, which says nothing about the file itself — download it and check in a player that can.

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Need to match a streaming loudness target instead? Use the Loudness Normalizer, then convert the result here.

What this does

This tool changes the container and encoding of an audio file: WAV, FLAC or MP3 out, at the sample rate and bit depth you choose. It does not touch the sound in any other way — no level change, no EQ, no dynamics. Converting at the same rate and depth to the same format gives you back the same audio.

Everything runs in your browser. Your file is decoded by the browser itself, converted by WebAssembly compiled from the reference encoders (LAME for MP3, libFLAC for FLAC), and handed straight back to you as a download. Those libraries and their licences are credited on the licences page.

Which format to choose

Format Lossless Use it for
WAV Yes DAW work, mastering, anything that will be processed again. Largest.
FLAC Yes Archiving and distribution where quality must be exact. Roughly half of WAV.
MP3 No Sharing, previews, anywhere size matters more than fidelity.

FLAC is the one worth knowing about: it is compressed and exact. Decoding a FLAC returns the original samples bit for bit, so the only cost against WAV is the time spent compressing. If you are keeping a master, keep it as FLAC or WAV, never MP3.

Sample rate and bit depth

Sample rate is how many times per second the audio was measured; it sets the highest frequency the file can represent. 44.1 kHz is CD and most streaming, 48 kHz is standard for video. Bit depth is how precisely each of those measurements is stored, which sets the noise floor: 16-bit is right for a finished file, 24-bit is what DAWs work in.

Both default to same as source here, and that is usually the right answer. Change them when a target spec demands it — a video timeline at 48 kHz, a distributor asking for 16-bit — not as a general improvement.

Converting to a higher rate or depth is offered because specs sometimes require it, but it adds no information. Upsampling a 44.1 kHz file to 96 kHz makes a bigger file containing exactly the same audio; it does not recover detail that was never recorded.

Limitations, honestly