MixMasterAIMP3 to WAV Converter
MP3 is a lossy format — small files, but with audio detail thrown away during encoding. WAV is uncompressed PCM. Converting an MP3 to WAV doesn't put the lost detail back; it just hands you the same audio in a container that DAWs, hardware samplers, and mastering chains accept directly.
How it works
Drop the MP3
Drop a file or click to choose. The conversion runs locally — your audio decodes in the browser via the Web Audio API and ffmpeg.wasm.
MP3, AAC, M4A, OGG, FLAC, AIFF all decode the same way
Pick bit depth and sample rate
16-bit / 44.1 kHz matches CD and is the right choice for distribution masters. 24-bit / 48 kHz gives more headroom and is the standard for DAW work — pick this if you're going to mix or master the result.
Codec: pcm_s16le or pcm_s24le · standard RIFF WAV
Download
ffmpeg writes a clean WAV with no re-compression. The audio inside is exactly what was in the MP3 — same artifacts, same bandwidth — but in a format every DAW will import.
Output is bit-identical to the decoded MP3 audio
Frequently asked questions
Will the WAV sound better than the MP3?
No. MP3 throws away audio data during encoding and that data isn't stored anywhere. Converting MP3 to WAV gives you the decoded audio in a different container — the artifacts and bandwidth limits of the source MP3 are still there. If you want lossless audio, you need a lossless source.
When should I convert MP3 to WAV?
Three common cases: (1) you're importing into a DAW that imports WAV faster or more reliably than MP3, (2) you're sending the file to a mastering engineer or distributor that requires WAV, (3) you're going to chop or process the audio and don't want a second generation of MP3 compression on top of the first.
What's the difference between 16-bit and 24-bit WAV?
Bit depth is the dynamic range. 16-bit covers 96 dB — fine for finished masters where everything sits well above the noise floor. 24-bit covers 144 dB and gives 48 dB more headroom, which matters when you're mixing or mastering and might push gain around. For distribution, use 16-bit. For further work, use 24-bit.
What sample rate should I pick?
Match the source. Most MP3s are 44.1 kHz; converting to 48 kHz won't add information, just resamples. Pick 48 kHz only if you're matching a video project. 96 kHz is for high-resolution mastering and rarely useful for an MP3 source.
Why is the first conversion slow?
ffmpeg.wasm downloads its core (~25 MB) the first time you convert anything in a session. After that it's cached — next conversion starts instantly until you clear browser cache.