Convert MP3 to WAV
Decode MP3 to uncompressed WAV at 16, 24 or 32-bit for editing, sampling and DAW import. Batch queue, local, and no size cap.
- Runs in your browser
- Files are never uploaded
- Free, no account
First use downloads the 10 MB video engine from this site. After that it is cached and works offline. It is WebAssembly served by onthisdevice.com — never a third-party CDN — and your file is still processed entirely on your own device.
What decoding to WAV does and does not do
A WAV file is a header followed by raw PCM samples. Converting an MP3 to WAV runs the MP3 decoder and writes out the samples it produces — nothing more. The result is a faithful, lossless copy of the decoded audio, which is not the same thing as a copy of the original recording. Everything the MP3 encoder discarded stays discarded.
So this conversion does not improve quality, and any tool claiming otherwise is wrong. What it gives you is a format that editors, samplers, DAWs and analysis tools can open, scrub and process directly, without re-decoding on every operation and without losing more quality each time you save.
Choosing a bit depth
| Depth | Bytes per sample | Stereo 44.1 kHz size | Use it when |
|---|---|---|---|
| 16-bit PCM | 2 | 10.1 MB per minute | Playback, burning, maximum compatibility. The CD standard. |
| 24-bit PCM | 3 | 15.1 MB per minute | The file is going into a mix where you will apply gain, EQ and effects. |
| 32-bit float | 4 | 20.2 MB per minute | Your DAW works in float and you want processing headroom that cannot clip. |
Converting a 16-bit source to 24-bit adds no information — the extra bits are zeros. It is still sometimes the right choice, because it means every subsequent processing step inside your editor has room below the noise floor to work in.
Working out the file size in advance
Uncompressed audio has no surprises. The formula is exactly:
bytes = sample rate × channels × (bit depth ÷ 8) × seconds
A four-minute stereo track at 44 100 Hz and 16-bit is 44 100 × 2 × 2 × 240 = 42 336 000 bytes, about 40 MB. Note that the source MP3’s bitrate does not appear anywhere in that calculation: a 128 kbps MP3 and a 320 kbps MP3 of the same track produce identically sized WAV files. This is also the reason upload-based converters dislike WAV output, and why several of them cap it.
The gap of silence at the start
MP3 is a frame-based format: the encoder pads the beginning of the stream so the first real audio starts on a frame boundary, typically 576 or 1105 samples in. Decoders that ignore this reproduce the padding as a few milliseconds of silence.
Most files written since the mid-2000s carry a LAME or iTunes gapless information tag describing exactly how much to trim, and FFmpeg honours it. Older or oddly encoded files may not, in which case a handful of milliseconds remain — trivial for most purposes, worth nudging by hand if you are chopping the file into precisely timed samples.
What to do with the WAV once you have it
- Sampling and looping. Samplers want raw PCM so they can loop at exact sample positions without decode latency.
- Editing. Cutting and re-saving an MP3 re-encodes it every time, compounding loss. Edit in WAV, export once at the end.
- Voice-over and video work. Most NLEs prefer PCM for scrubbing performance and frame-accurate sync.
- Analysis and machine learning. Nearly every audio library expects PCM input and would decode the MP3 itself anyway.
- Legacy hardware. Some announcement systems, telephony platforms and museum installations read WAV and nothing else.
Big files, no cap
A one-hour podcast becomes roughly a 600 MB WAV. That single fact is why so many online converters refuse this job or quietly downsample it — they would have to receive the MP3, hold the WAV, and send it back. Here the decode happens in FFmpeg compiled to WebAssembly inside your browser tab, so the file never travels and the only limit is your device’s memory. The browser sandbox tops out around 2 GB of working memory, which is about three hours of stereo 16-bit audio.
Going the other way
To shrink a WAV back down — or to compress recordings your DAW exported — use WAV to MP3, which shows the exact size ratio for each file. For a lossless archive that is roughly half the size of WAV, convert to FLAC in the general audio converter instead.
Questions
Does converting to WAV improve the quality?
No. WAV is a lossless container, so it faithfully preserves whatever the MP3 decoded to — including everything the MP3 encoder already threw away. What WAV gives you is a format that editors, samplers and DAWs can scrub and process without re-decoding, and one that will not lose more quality on each save.
Which bit depth should I pick?
16-bit for anything you will simply play or burn — it is the CD standard and the most compatible. 24-bit if the file is heading into a mix where you will apply gain and effects. 32-bit float if your DAW works in float and you want headroom that cannot clip during processing.
How big will the WAV be?
Exactly sample rate × channels × bytes per sample × seconds. Stereo 44.1 kHz 16-bit is 10.1 MB per minute, so a four-minute track is about 40 MB regardless of the MP3 bitrate it came from. 24-bit is 50% larger again. This is the file size upload converters refuse to accept.
Why does the WAV start with a moment of silence?
MP3 encoders add padding at the start of the stream, and the decoder faithfully reproduces it. Most modern files carry a LAME or iTunes gapless tag that FFmpeg uses to trim it. If a few milliseconds remain, nudge the region start in your editor.