MP3 → WAV

No, and it is worth being blunt: nothing can. The MP3 already discarded detail during its original encode, and wrapping what remains in a lossless WAV preserves exactly that — no more. The file gets several times larger and sounds identical. It is still the right move before editing, because WAV will not add a second generation of artefacts when you export. WAV is the correct format for editing, mastering and archiving, because it can be re-encoded any number of times without accumulating artefacts. MP3 is technically outclassed by AAC and Opus at every bitrate, cannot store more than two channels usefully, and always adds a small encoder delay.

Not intact. MP3 carries a proper metadata block, but WAV has no standard place for artist, album, track number or cover art, so tags are dropped rather than mangled. Keep the MP3 as the tagged copy, or re-tag afterwards with a tool like Mp3tag or MusicBrainz Picard. WAV is enormous — roughly 10 MB per stereo minute at CD quality — and the classic format has a 4 GB ceiling and no room for proper tags.

Default 192 kbps for lossy WAV; pass-through for lossless WAV. Override to 320 kbps if you want maximum lossy fidelity, 128 kbps for size-constrained podcast distribution, or 96 kbps for voice-only sources where the smaller file matters more than studio detail.

If MP3 is lossy (MP3, AAC) and WAV is lossless (WAV, FLAC), the WAV is no better than the MP3 — you cannot recover information already discarded by the lossy MP3 codec. If MP3 is lossless and WAV is lossy, expect the WAV encoder to recompress; at 192 kbps the loss is imperceptible to most ears.

Yes — title, artist, album, year, track number, album art, BPM, replay-gain are read from the MP3 container and written into the equivalent fields on the WAV container. ID3v2 (MP3), Vorbis comments (OGG, FLAC), MP4 atoms (AAC, M4A), RIFF INFO (WAV), iXML chunks all map cleanly. WAV is Microsoft and IBM’s RIFF-based audio container from 1991, and it is what almost every audio interface and DAW writes on record.

Yes — drop a folder of MP3 files in and we process them in parallel. Premium has more parallel workers and no per-file size cap, so a 500-file batch finishes in minutes rather than tens of minutes. Folder structure is preserved in the output ZIP.

By default yes (a 48 kHz MP3 produces 48 kHz WAV, a 44.1 kHz MP3 produces 44.1 kHz WAV). For specific compatibility — e.g. downsampling 96 kHz studio masters to 44.1 kHz WAV for CD burning, or upsampling 22 kHz voicemail to 44.1 kHz WAV — the sample-rate dropdown applies high-quality SOX-style resampling.

Yes — the loudness-normalize option applies ITU-R BS.1770 / EBU R128 normalization to the WAV output, targeting -14 LUFS (streaming standard) or -16 LUFS (podcast standard) or a custom value. Useful when batch-converting tracks with varying mastering levels into a single WAV playlist.

MP3 plays universally on every device made in the last 20 years. AAC plays on Apple, most Android, Sonos, and modern car stereos. FLAC plays on Sonos and recent Android, less well on older iPods. WAV plays on everything but the file is huge. The device-target dropdown picks a safe WAV codec for the platform you specify.

Yes — uploaded MP3 files are processed in isolated workers and deleted within minutes. We never play, store, or share the audio content. The pipeline runs on hardware we control end-to-end; no third-party services receive your file.

Same-codec re-mux (e.g. AAC inside M4A -> AAC inside MP4): 10-30 seconds. Codec change (e.g. FLAC -> MP3 or WAV -> AAC): typically 10-20% of source duration, so a 1-hour MP3 -> WAV finishes in 6-12 minutes on the standard pipeline.