WAV is the uncompressed audio format required by many professional audio tools, hardware samplers, DJ controllers, and video editors. This converter decodes your MP3 and outputs a 16-bit, 44.1 kHz stereo WAV file — the standard format accepted by virtually every audio application and device. Runs in your browser via FFmpeg WebAssembly; your audio files never leave your device.
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Up to 200.0 MB per file
Upload your MP3 file
Drop an MP3 file onto the converter. Any MP3 bitrate (64–320 kbps) is supported. Up to 200 MB input.
Convert
Click Convert. FFmpeg decodes the MP3 and outputs a standard PCM WAV file at 44.1 kHz, 16-bit stereo. No quality settings needed — WAV is always uncompressed.
Download the WAV
The WAV file is significantly larger than the original MP3 (typically 10× for a 128 kbps MP3) because it stores uncompressed audio. That's expected and correct.
Use in your software
Import into Audacity, GarageBand, Logic Pro, Pro Tools, Ableton Live, or any DAW. Load into a hardware sampler, drum machine, or DJ controller that requires WAV format.
No. MP3 uses lossy compression — once audio data is discarded during MP3 encoding, it cannot be recovered. Converting to WAV creates an uncompressed copy of the current (already-lossy) audio. Think of it as a lossless copy of a compressed original. The WAV won't sound better than the MP3 — it just guarantees no further quality loss when re-edited or re-exported.
Common reasons: (1) Hardware devices (samplers, drum machines, car stereos, game consoles) often only read WAV because decoding MP3 requires extra CPU/DSP. (2) DAWs prefer WAV for editing because MP3 frame boundaries don't align perfectly with project sample rates, causing clicks during loop editing. (3) Video editors may require WAV for frame-accurate audio sync.
44.1 kHz sample rate and 16-bit depth — the standard CD quality. This is compatible with virtually all audio software and hardware. Some professional applications prefer 48 kHz (broadcast standard) or 24-bit depth; use a desktop tool like Audacity or SoX for those specific requirements.
A 128 kbps MP3 compresses audio to roughly 1 MB per minute. An uncompressed 44.1 kHz 16-bit stereo WAV is ~10 MB per minute. This is expected — the file size increase is the "cost" of uncompressed audio. A 5 MB MP3 might become a 50 MB WAV.
Yes, with the caveat that the source quality is limited by the original MP3 encoding. For professional production starting from MP3 source material, 44.1 kHz 16-bit WAV is perfectly suitable. If you have access to the original lossless source (FLAC, WAV, AIFF), always use that instead of an MP3-derived WAV.
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