MP3 to WAV Converter

About the MP3 to WAV Converter

Turn an MP3 — or an M4A, OGG, Opus, FLAC, AIFF or the soundtrack of a video — into an uncompressed WAV file that every audio editor, DAW, hardware sampler, DJ program and CD-burning app can open. Choose 16-bit or 24-bit PCM or 32-bit float, keep the sample rate or set 44.1 kHz (CD) or 48 kHz (video), and keep the channels or make it stereo or mono.

The conversion is gapless: a decoded MP3 normally has about 25 milliseconds of silence in front of the sound and some padding at the end. When the MP3 carries a LAME header (as files from LAME, FFmpeg and most apps do), this page cuts those off, so the WAV holds exactly the samples that were encoded — loops stay seamless and tracks line up in a timeline.

If you like, the level can be set on the way: the highest peak at −1 or −0.1 dBFS, or a loudness of −14, −16 or −23 LUFS, measured as ITU-R BS.1770 describes. Everything runs in your browser; the file is never uploaded. Without a pass, the WAV is a free preview: its first half (up to 30 seconds) with a short MySmartCoPilot chime every 10 seconds.

Which settings to choose

• Audio CD: 16-bit, 44.1 kHz, stereo — the format of audio CDs (the Red Book standard, IEC 60908). Burning apps want exactly this.
• Editing & DAWs: 24-bit at the file’s own sample rate. The extra bits keep the decoded sound without rounding it, so later gain changes, fades and effects start from the best copy.
• 32-bit float: keeps peaks above full scale. Loud MP3s often decode slightly past 0 dBFS; 16- and 24-bit files must cut those peaks off, float keeps them so you can lower the level later without distortion.
• Sample rate: “Same as the file” is right unless the program you use asks for 44.1 or 48 kHz. Changing it uses a band-limited filter, so no false tones fold back into the sound.
• Dither (16-bit only): rounding to 16 bits adds a little noise; triangular (TPDF) dither of one step makes that noise smooth and even instead of tied to the music, which keeps quiet fades clean. It is added only where the values are not already whole 16-bit steps.

Gapless: why a plain decode makes the WAV longer

An MP3 encoder delays the sound by a few hundred samples (576 in LAME) and pads the end to fill the last frame, and the decoder adds its own delay — 529 samples in Chrome’s decoder, as in FFmpeg’s. A WAV made by simply decoding the MP3 therefore starts about 25 ms late and runs on after the end.

The LAME header (in the first frame of the MP3, written by LAME, FFmpeg and most apps) records the delay and padding. This page reads it, measures your browser’s decoder delay by decoding a tiny test MP3 with a click at a known sample, and cuts exactly that much, so the WAV has the length the source had before it was encoded. You can turn this off to keep the decoder’s raw output.

How big the WAV will be

WAV stores every sample uncompressed: channels × sample rate × bytes per sample.

• 16-bit, 44.1 kHz, stereo: 1,411.2 kbit/s, about 10 MB a minute
• 24-bit, 48 kHz, stereo: 2,304 kbit/s, about 16.5 MB a minute
• 32-bit float, 48 kHz, stereo: 3,072 kbit/s, about 22 MB a minute

A 320 kbps MP3 is about 2.3 MB a minute, so the WAV is four to ten times larger. A WAV file cannot pass 4 GB; longer ones are written as RF64 (EBU Tech 3306), the large-file form of WAV that current editors read.

Standards behind the file

The file follows Microsoft’s RIFF/WAVE format: mono and stereo PCM use the plain WAVE_FORMAT_PCM header that hardware samplers and CD apps read, 32-bit float uses WAVE_FORMAT_IEEE_FLOAT, and more than two channels use WAVE_FORMAT_EXTENSIBLE with the usual speaker order. Large files use RF64. MP3 is defined in ISO/IEC 11172-3 and 13818-3; loudness is measured as in ITU-R BS.1770.

Limitations

• Converting to WAV cannot bring back what MP3 compression removed: the WAV is an uncompressed copy of the decoded MP3 (rounded to whole 16-bit steps if you choose 16-bit), not of the original recording.
• Cover art and lyrics stay behind: WAV has no standard place for them. Title, artist, album, year, genre, comment and track number are kept as RIFF INFO tags.
• One file at a time. Very long files take time and memory, especially on phones; the WAV is built in the browser before it is downloaded.
• Copy-protected files (DRM, such as old iTunes .m4p purchases) and codecs browsers have no decoder for (WMA, AMR, APE) cannot be converted.
• Gapless trimming needs the LAME header; MP3s without one are decoded as they are (the page says so).

Privacy

Everything happens in your browser. What you enter or open here is not uploaded or stored by MySmartCoPilot. The audio engine downloads from MySmartCoPilot the first time you open a file (a WebAssembly decoder too, only for a file your browser cannot decode itself). Your audio is never uploaded.