MP4 to WAV Converter

Extract the audio from an MP4 as uncompressed WAV, in your browser, at the original sample rate. This is the format transcription tools and editors actually want.

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 MP4 to WAV

  1. 1Drop your .mp4 onto the box above, or click to pick a file.
  2. 2The audio track is decoded and written straight out as WAV. There is no quality setting, because uncompressed audio has none.
  3. 3Play it here to confirm, then download. WAV files are large, so expect the download to be much bigger than the MP3 equivalent.

The case for uncompressed

Most people who convert MP4 to WAV are not planning to listen to the result. They are feeding it to something: Whisper or another speech model, a forced aligner for subtitles, a DAW for a rough edit, a script that measures loudness. All of those work on samples, and all of them will decode a compressed file before they start.

Handing them an MP3 therefore adds a lossy encode in the middle of your chain and buys nothing. Better to extract WAV from MP4 once, work on that, and compress at the end if a compressed deliverable is what you need.

What it is, and what it is not

The audio track in an MP4 is virtually always AAC, which is lossy. Decoding it to WAV preserves exactly what that AAC decodes to, which means every artifact the original encode introduced comes along. Nothing is restored. There is no version of this operation that recovers the studio recording, and any tool suggesting otherwise is selling something.

What you do get is a stopping point: from here on, nothing further is lost. Cuts, fades, normalisation and analysis all operate on plain integers, and you decide when and whether a lossy encode happens.

Size, and the ceiling

Uncompressed audio is expensive. Stereo at 44.1 kHz and 16 bits is about 10 MB per minute, so the numbers get uncomfortable quickly. A five minute clip is 50 MB. A half-hour interview is 300 MB.

Input files up to 500 MB are accepted. Above that a browser tab cannot hold the container, and being told so before you start beats having the tab killed. For long material, trim the section you need first.

Everything happens on your machine. The engine is ffmpeg compiled to WebAssembly, downloaded once and cached by your browser afterwards.

Frequently asked questions

Why WAV instead of MP3?
Because you are handing the audio to software rather than to your ears. Speech recognition models, forced-alignment tools, DAWs and audio analysis code all want plain samples. Giving them an MP3 means they decode it anyway, and any tool that re-encodes in the middle of your chain costs you quality for no benefit.
Is this better quality than converting to MP3?
It avoids a second lossy pass, which is a real advantage. But the audio inside an MP4 is almost always AAC, so it has already been compressed once. The WAV is a faithful copy of that decoded AAC, not a recovery of the original recording. Faithful is the most any tool can offer here.
How big will the file be?
About 10 MB per minute of stereo at 44.1 kHz, and about 5 MB per minute for mono. A 30-minute video therefore produces roughly 300 MB of WAV. That is normal, and it is why WAV is a working format rather than a delivery format.
Can I do a whole hour?
The input limit is 500 MB, which most hour-long videos exceed, and the output would be around 600 MB. For something that long, split the video first or use a desktop tool. This is a browser tab, and it says so rather than crashing.
Does it resample to 48 kHz?
No. The source rate is kept exactly, which matters because some transcription pipelines assume the header tells the truth. If a model needs 16 kHz mono, do that resample in the pipeline that needs it, where it is a documented step.

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