M4A to WAV

Convert M4A to WAV in your browser — no upload, no queue, no size cap set by somebody else's server. Real PCM output, 16- or 24-bit. No upload — runs 100% in your browser

Drop your M4A file here
or click to browse
100% private: the file is processed in your browser and never uploaded anywhere.

How to use this tool

  1. Drop your .m4a file on the box above, or click to browse.
  2. Choose the bit depth and sample rate you need (defaults are 16-bit / 44,100 Hz).
  3. The file is decoded by your own browser and written as uncompressed PCM.
  4. Click Download. Nothing was uploaded at any point — you can disconnect the network after this page loads.

What actually happens when you convert M4A to WAV

An M4A file is AAC audio inside an MP4 container — the format Apple's Music app, iTunes downloads and iPhone voice memos default to. AAC reaches its small file size by discarding parts of the signal it judges hard to hear, so the data on disk is already a reconstruction rather than a recording.

Conversion runs in two steps. Your browser's own AAC decoder expands the file back into PCM — the raw sequence of sample values a sound card would play — and this page then writes those values into a WAV container with nothing further done to them. WAV carries no compression at all: it stores the numbers as they are. That means the step from decoded audio to WAV adds no new damage, which is why WAV is the working format for editors, DAWs, CD-burning tools and broadcast hand-off.

It also means what people usually hope for does not happen. A WAV made from an M4A is never better sounding than the M4A — the detail AAC threw away is gone, not restored. What you gain is compatibility and editability, not fidelity. If the source truly matters, re-export from the original session or a lossless master (FLAC/ALAC/WAV); converting a lossy file again only locks in what is missing.

What it costs you in file size

Uncompressed means exactly that. Stereo audio at 44,100 Hz and 16-bit needs 2 bytes per sample, two channels, 44,100 times a second — 1,411.2 kbit/s, which works out to 10,584,000 bytes per minute (about 10.09 MiB). A three-minute track becomes roughly 30 MB; a one-hour recording becomes roughly 606 MB. Choose 24-bit instead and the same audio costs 2,116.8 kbit/s, half again larger.

SettingBit rateSize per minute (stereo)
44,100 Hz, 16-bit1,411.2 kbit/s10.09 MiB
48,000 Hz, 16-bit1,536 kbit/s10.99 MiB
44,100 Hz, 24-bit2,116.8 kbit/s15.14 MiB
48,000 Hz, 24-bit, mono1,152 kbit/s8.24 MiB

Choosing bit depth and sample rate

16-bit at 44,100 Hz is the CD standard and the safe default: every editor, every CD burner, every phone plays it. 24-bit lowers the noise floor by roughly 6 dB for each extra bit — theoretically about 96 dB of dynamic range at 16-bit versus about 144 dB at 24-bit — which matters only if you plan to process the audio further, since headroom during mixing is where those extra bits earn their keep.

One honest limitation: the Web Audio API resamples during decoding to whatever rate the audio context uses, and the original sample rate of your file is not readable from the page. So you pick the output rate explicitly. If your source is 48,000 Hz and you pick 44,100 Hz, one resampling pass happens — clean, but not bit-exact. If you need the source rate preserved exactly, use ffmpeg -i input.m4a output.wav locally instead.

Frequently Asked Questions

Does converting M4A to WAV improve the sound quality?

No. The AAC audio inside an M4A is lossy — parts of the signal were already discarded to keep the file small. Converting produces uncompressed PCM, which stops any further loss, but it cannot bring back detail that is not there. Anyone who hears an improvement is usually hearing a different playback volume, not recovered detail.

Is my file uploaded anywhere?

No. The decoding is done by your browser's own audio engine and the WAV is assembled by JavaScript in this page. The file never leaves your device — you can turn off your network connection after the page loads and it still works.

Why did the conversion fail?

The two usual reasons are DRM and a mislabelled file. Music bought from the iTunes Store before 2009 was often protected, and protected files cannot be decoded by anyone without Apple's authorisation. The other case is a file renamed to .m4a that is not actually AAC audio. Both produce an explicit error here rather than a broken download.

What bit depth should I use?

Use 16-bit unless you have a reason not to: it is the CD standard, everything opens it, and it is already beyond the dynamic range of real recordings. Pick 24-bit only when the file is going into further processing — mixing, heavy EQ or noise reduction — where the extra headroom protects you from rounding artefacts.

Which sample rate should I choose?

44,100 Hz for music and CD delivery, 48,000 Hz for audio that will sit alongside video. Note that your browser resamples during decode to whichever rate you select, so if you must preserve the file's original rate exactly, use a local tool such as ffmpeg rather than this page.

Can I convert several files at once?

Not on this page — each conversion holds the full decoded track in your browser's memory (roughly 20 MiB per minute of stereo audio), so running them in parallel would risk crashing the tab. Convert them one at a time; each download replaces the previous one in memory immediately.

How large a file can I convert?

Whatever your device's memory allows. The practical ceiling is the decoded PCM, which is about four times the size of the compressed source: a one-hour stereo recording needs roughly 1.2 GB of memory before it can be written out. Files over 500 MB are refused up front to save you the wait.

Does a WAV file keep metadata such as artist and album art?

This converter writes a plain PCM WAV with no metadata chunks, so tags and artwork are dropped. If you need them preserved, carry them in your library software's sidecar files, or use a tool that writes ID3/INFO chunks rather than this page.

Can I convert to mono to halve the size?

Yes — tick 'Mix down to mono'. The two channels are averaged sample by sample, which halves the file size and is what you want for speech, podcasts and voice memos. It is not reversible from this page, so keep the original stereo file.

Related Tools

OGG to WAV