MP3 → WAV
MP3 to WAV converter
The MP3 is decoded to uncompressed PCM and written as a WAV. The file gets about 8× larger, and the audio quality is exactly what the MP3 already was — decoding restores nothing.
…or drag files and folders here. Add as many as you like — everything is processed on this device.
.aac .m4a .mp3 .oga .ogg .opus
In one paragraph
Plenty of software refuses MP3 on purpose: hardware samplers, Pro Tools sessions on older versions, broadcast playout systems, phone IVR platforms, some Unity import pipelines and a lot of scientific analysis tools. All of them want PCM in a WAV.
This is the one conversion where it is important to be clear about what you are getting. Decoding an MP3 gives you the MP3’s audio in a lossless wrapper. The 16 kHz shelf, the smeared cymbals and the pre-echo the encoder introduced are all still there, now stored at 1,411 kbps instead of 192.
Does this change the audio quality?
Lossless in the direction it can be: every sample the MP3 decoder produces is written exactly. But converting a lossy file to a lossless format does not recover the discarded information. A WAV made from a 128 kbps MP3 sounds like a 128 kbps MP3.
- The WAV will be roughly 7–11× the size of the MP3. A 4 MB MP3 becomes about 40 MB of WAV.
- MP3 decoders introduce a small amount of padding, so the WAV may be a few milliseconds longer.
- ID3 tags do not survive into WAV in a way most software reads. Track names are best kept in the filename.
A worked example
Preparing MP3 samples for a hardware sampler that needs 16-bit WAV
- Drop the MP3s in. The source panel shows each file’s sample rate and channel count.
- Set Bit depth to 16 and Sample rate to 44100 — most hardware samplers insist on both.
- Set Channels to Mono for one-shots so you are not wasting sampler memory on a duplicated channel.
- Convert all and download the ZIP, then copy the WAVs to the sampler’s card.
16-bit unless something downstream needs more
An MP3 decodes to values that fit comfortably in 16-bit. Choosing 24-bit or 32-bit float makes the file 1.5–2× larger and adds precision the source never had — worth it only if you are about to apply gain, EQ or time-stretching in a DAW and want headroom for the processing chain. Keep the source sample rate; resampling is another small, avoidable quality cost.
Supported formats, limits and browser notes
Sonic Batch decodes with the browser when it can (faster, no engine download) and falls back to the self-hosted WebAssembly engine when it cannot. Every combination below works in every listed browser; the column only tells you which decoder does the work.
| Input | What it is | Chrome | Firefox | Safari | iOS |
|---|---|---|---|---|---|
| MP3 | The universal lossy format. Plays on essentially every device made since 1998. | Native | Native | Native | Native |
| AAC | Bare AAC frames with no MP4 wrapper — used by broadcast streams and some hardware. | Native | Partial | Native | Native |
| M4A | Apple’s default AAC container. Better quality than MP3 at the same bitrate. | Native | Native | Native | Native |
| OGG | Royalty-free lossy codec. Standard in game engines, Godot, Unity and Wikimedia. | Native | Native | Engine | Engine |
| Opus | The best-sounding lossy codec below 128 kbps. Used by WhatsApp, Discord and YouTube. | Native | Native | Partial | Partial |
| Output | Encoder | Bitrates / depth | Sample rates | Tags |
|---|---|---|---|---|
| WAV | pcm_s16le | 16, 24, 32-bit | 8–96 kHz | Yes |
- Max per file
- 400.0 MB
- Max per queue
- 4.00 GB
- Max duration
- 4:00:00
- Max archive
- 1.50 GB
These are enforced ceilings, checked before any decoding starts, not measured maximums — a file over the limit is rejected rather than allowed to exhaust the tab’s memory. Real-world capacity is lower on older phones, which is why the scheduler runs one file at a time on iOS and Safari.
Where your audio goes
Nowhere. Your file is read from disk into this browser tab with the File API, decoded and encoded by code running in that same tab, and handed back to you as a download. It is never transmitted.
That is a structural property of the application, not a policy promise: there are no API routes, no server actions and no third-party processing services in this project, so there is nothing capable of receiving a file. The page’s Content-Security-Policy also restricts connections to this origin, and the automated test suite includes a browser test that fails if any network request during import, preview, conversion or download carries media bytes.
The trade-off, stated plainly:results live in your tab’s memory until you download them. Closing or reloading the page discards them. There is nothing on a server to recover, because nothing was ever put there.
Troubleshooting
The WAV is enormous.
That is what uncompressed means: about 10 MB per stereo minute at 44.1 kHz/16-bit. If size matters and your software supports it, FLAC is lossless too and roughly half the size.
My DAW says the WAV has a different length than the MP3.
MP3 is frame-based and the decoder pads the final frame. Differences of a few tens of milliseconds are normal; anything over 250 ms is flagged in the manifest.
Can I get the original CD quality back?
No. If the MP3 came from a CD, the discarded information is gone. You need the original disc or a lossless rip.
Frequently asked questions
Does MP3 to WAV improve quality?
No. It changes the storage format, not the audio. Any tool that claims to restore quality by converting to a lossless format is misleading you.
Why would I do this conversion at all?
Compatibility and processing. PCM is what audio software wants to work with, and some hardware and platforms will not accept anything else.
Which bit depth should I choose?
16-bit for playback and sampling, 24-bit if the file is going into a mixing session, 32-bit float only if your DAW works in float and you want to avoid any clipping during processing.
Can I batch a whole album?
Yes, though watch the total size — an album of WAVs is around 600 MB. Sequential download mode saves each file as it finishes rather than assembling one large ZIP in memory.