MP3 to WAV Converter

Decode an MP3 into an uncompressed WAV in your browser, at the original sample rate. Useful for software that only reads WAV, and honestly useless for quality.

The converter is ffmpeg compiled to WebAssembly. It downloads once (30 MB) when you pick your first file, then stays in your browser cache.

Your file is processed by your own browser and never uploaded. The engine itself is a one-time 30 MB download the first time you use it, and after that it comes out of your browser cache.

How to use MP3 to WAV

  1. 1Drop your .mp3 files onto the box above, or click to pick them.
  2. 2Each file is decoded and written out as a WAV. There is no quality setting, because uncompressed audio has none.
  3. 3Play the result to check it, then download. A batch comes out as one .zip.

What this actually does

An MP3 is a compressed representation of audio. To play it, any device has to decode it back into ordinary samples first. To convert MP3 to WAV is simply to stop after that step and write those samples to disk instead of sending them to your speakers.

So the WAV is not a better version of the MP3. It is the MP3, decoded, in the form your sound card was going to receive anyway. Every compression artifact the encoder introduced is still there, faithfully preserved in uncompressed form.

The legitimate reasons

There are plenty, and none of them are about fidelity:

  • Hardware samplers and older drum machines that read WAV from a card
  • CD authoring software, which wants 44.1 kHz PCM
  • Game engines and audio middleware that require uncompressed assets
  • Transcription, forced alignment and analysis tools with a WAV-only input
  • Broadcast and podcast delivery specs that mandate PCM
  • Any editing step where you do not want a re-encode in the middle of the chain

That last one is the most useful case. If you are going to chop, fade and process an MP3 and then export it again, decoding once to WAV up front means the editing happens on plain samples rather than being re-encoded repeatedly.

What comes out

Standard RIFF WAV, 16-bit signed PCM, at the source sample rate and channel count. A mono MP3 produces a mono WAV rather than two identical channels. That is what every DAW, sampler and analysis tool expects, and it is the format that will not raise a question anywhere.

Expect the file to be roughly ten times larger. Files up to 500 MB are accepted as input, though the output can comfortably exceed that, so long recordings are better split first. Everything runs on your own machine.

Frequently asked questions

Does converting MP3 to WAV improve the sound?
No, and it is worth being blunt about it. The WAV contains exactly what the MP3 decoded to, artifacts included. Nothing is restored, because the information the MP3 encoder discarded is not stored anywhere in the file. What you get is a bigger file with identical audio.
Then why would I do it?
Because something downstream demands WAV. Hardware samplers, CD burning software, older DAWs, some transcription and forced-alignment tools, game engines that want uncompressed assets, and a fair number of broadcast delivery specs all refuse compressed input. That is a real need, and it has nothing to do with quality.
How much bigger will it get?
Around ten times, at typical MP3 bitrates. A 5 MB MP3 becomes roughly 50 MB of WAV. Uncompressed stereo at 44.1 kHz costs about 10 MB per minute regardless of what the source was.
Why 16-bit and not 24-bit?
Because 24-bit would be padding. MP3 does not store integer samples, so the extra bit depth would carry no additional information from the source: the file would be 50 percent larger and contain the same audio. If a specific tool of yours requires 24-bit, do that conversion in the tool itself, where the rest of the chain is also 24-bit.
Is the sample rate changed?
No. A 44.1 kHz MP3 produces a 44.1 kHz WAV. This is the reason the tool uses a real decoder rather than the browser's Web Audio API, which resamples everything to whatever your output device runs at.

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