WAV Converter
Produce uncompressed PCM WAV files at 16, 24 or 32-bit float — the format every audio editor, sampler and video editor accepts without argument.
Convert with this tool
or choose a file from your device — it is processed in this tab and never uploaded
Accepted: MP3, FLAC, M4A/AAC, OGG Vorbis, Opus, AIFF, MP4/MOV audio tracks
Files stay on your device. There is no upload, no queue and no watermark, so a conversion works even with the network switched off after the page has loaded.
Supported formats & processing
| Accepted input | MP3, FLAC, M4A/AAC, OGG Vorbis, Opus, AIFF, MP4/MOV audio tracks |
|---|---|
| Produced output | WAV — 16-bit PCM, 24-bit PCM or 32-bit IEEE float |
| Processing | Web Audio API + canvas + native JavaScript encoders |
| Upload required | No — the file is read locally and never transmitted |
How to use this tool
- Add the file you want in WAV formThe browser decodes it to PCM at the source sample rate.
- Choose the bit depth16-bit matches CD audio and is fine for playback; 24-bit gives headroom for editing and mixing; 32-bit float preserves values above 0 dBFS.
- Set the sample rateKeep original unless a delivery spec requires 44.1 or 48 kHz.
- ConvertA short file finishes almost instantly because no psychoacoustic encoding is involved.
- Download the WAVExpect a large file: uncompressed stereo at 44.1 kHz runs about 10 MB per minute.
About this tool
WAV is not a codec, it is a container: a 44-byte RIFF header followed by raw PCM samples. That is why it is the interchange format of choice for editing. Every sample in your source is written out in full precision, with no quantisation step and no psychoacoustic model deciding what to throw away.
This converter writes a correct RIFF/WAVE header for 16-bit signed integer, 24-bit packed integer or 32-bit IEEE float data, with the channel count and sample rate you select. Because the browser decodes the source first, MP3, FLAC, M4A, OGG and Opus inputs all arrive as clean PCM before the WAV file is assembled.
Features
- True RIFF/WAVE headers written byte by byte, not a container guess
- 16-bit, 24-bit packed and 32-bit IEEE float output
- Sample rates from 8 kHz to 96 kHz
- Mono or stereo with correct block alignment and byte-rate fields
- Uncompressed output, so no generation loss on repeated conversions
- File-size expectation stated up front so nothing surprises you
Frequently asked questions
Why is the WAV file so much bigger than the MP3?
Uncompressed audio stores every sample of every channel. Stereo at 44.1 kHz and 16-bit is 1,411 kbps — roughly ten times a 128 kbps MP3 — so about 10 MB per minute of audio.
Which bit depth should I pick?
16-bit for listening and CD-style delivery, 24-bit when the file will be edited or processed further, 32-bit float when you are moving audio between digital workstations and want to preserve overs.
Does converting MP3 to WAV improve the sound?
It does not add detail, because the MP3 already discarded it. What it does give you is a file that will not lose anything further when you edit, slice or re-encode it.
Can I open the result in any editor?
Yes. The header is a standard RIFF/WAVE structure with a single fmt chunk and a single data chunk, which Audacity, Adobe Audition, Reaper, Logic and FFmpeg all read directly.