Convert M4A to WAV

Converting M4A to WAV produces uncompressed 16-bit PCM at the original sample rate, which is what a DAW, a sampler or a piece of hardware wants and what an M4A cannot give it. It does not improve the audio — unless the M4A holds Apple Lossless, in which case the WAV is a bit-exact copy of an untouched master rather than a larger container for damaged audio.

  • Where it runs In your browser. The file is never uploaded.
  • Lossy Some detail is traded for size. WAV cannot hold everything an M4A can.
  • File size limit Up to 100 MB per file, free, without an account.
  • Worth knowing The file gets much larger without getting better. What the original compression removed is gone; a lossless format can only preserve what is left.

Up to 100 files at once. Mixed formats are fine.

Find out whether your M4A is AAC or Apple Lossless first

M4A is a container and the registry lists two codecs inside it: AAC and ALAC. Which one you have decides whether this conversion is a compatibility step or an archival one, and it is the single most useful minute you can spend before converting anything.

Size answers it immediately. A four-minute AAC M4A is four to eight megabytes; the same track as Apple Lossless is twenty to thirty. On a Mac, Command-I on the file names the codec outright; in the Music app, adding a Kind column shows "Apple Lossless audio file" against "AAC audio file". If it is ALAC, the WAV you get is bit-for-bit the audio that went into the encoder, and this page's warnings about not restoring anything simply do not apply to you.

An AAC M4A cannot be improved by making it larger

For the usual case — AAC — the honest statement is that decoding to WAV restores nothing. AAC works by permanently discarding parts of the recording the ear is least likely to notice. Those parts are not compressed, not hidden and not recoverable; they are absent. Writing the surviving samples out uncompressed produces a file five to ten times larger that sounds precisely the same, because it contains precisely the same audio.

That does not make the conversion pointless. It makes the reason for it something other than quality — and the real reasons are good ones, which the next two sections are about. It only becomes a mistake when someone converts an entire library to WAV in the belief that they are upgrading it, and deletes the M4As afterwards.

Why a DAW would rather have the WAV

Some hosts refuse M4A outright. Many hardware samplers, grooveboxes, field recorders and DJ controllers read PCM off a card and contain no AAC decoder at all — the file is not rejected so much as invisible. Where a DAW does import an M4A, a good number of them quietly write a WAV into the project folder on import, which is this conversion happening without being asked for.

The other reason is behaviour rather than compatibility. A compressed file has to be decoded before it can be played, scrubbed or waveform-drawn, and a session with forty compressed clips in it is doing that work continuously. Uncompressed audio can be read straight off the disk at the byte offset the playhead is at, which is why scrubbing is instant and why studio workflows have used WAV since 1991 without seriously considering an alternative.

Editing without stacking a second round of damage

The second good reason is generational. Every time audio is exported back to a lossy format it is compressed again, and the losses accumulate — a clip that has been through five edit-and-export cycles as AAC is audibly worse than one that has been through one.

Working in WAV breaks that chain. Convert once, do the trimming, the level matching, the fades and the processing on uncompressed audio, and encode once at the end to whatever the delivery format is. The source is no better than it was, but it stops getting worse with each pass, which over a project of any length is the difference that shows.

The size arithmetic, and when it becomes a problem

Sixteen-bit stereo PCM at 44,100 Hz is 1,411 kilobits per second, which is 10.1 megabytes per minute. A four-minute track is about 40 MB, a 45-minute rehearsal recording is around 455 MB, and a two-hour interview is a little over 1.2 gigabytes. WAV offers no quality control because there is nothing to trade; that is what makes it a working format.

The 100 MB per-file ceiling here is the practical limit, and it corresponds to roughly ten minutes of stereo audio. For longer material, FLAC is the sensible target: it is lossless in exactly the same sense, typically compresses to half the size, and every audio editor made in the last fifteen years reads it. Reach for WAV specifically when a piece of software or hardware insists on it.

What the RIFF INFO chunk can and cannot hold

WAV metadata lives in a RIFF INFO chunk, and it is much smaller than what an M4A carries. Title, artist, album, track number, genre, date and comment map across. Album artist, disc number, description and lyrics have no INFO equivalent and are dropped, and so is embedded cover art — there is no picture field in the chunk at all.

There is a second, sharper limit worth knowing: RIFF INFO values are ISO 8859-1, so a title containing Japanese, Cyrillic, Greek or Devanagari characters cannot be represented and is left out rather than written as nonsense. For a session file none of this matters. For a music library it matters a great deal, and it is the strongest argument for FLAC over WAV as an archival target.

Multichannel M4A files and what WAV does with them

The registry records M4A as supporting up to 48 channels and WAV up to 18, so most surround material passes through intact — a 5.1 film soundtrack extracted as M4A converts to a six-channel WAV rather than being folded down, which is the outcome anyone doing post-production work wants.

The channel order is the thing to check on arrival. WAV files carry channel assignments in an extension chunk that not every program reads, and a six-channel file imported into a host that assumes a different order will put the centre channel somewhere unhelpful. Play the first few seconds through your monitoring before you build anything on top of it.

Encoder padding at the head of the file

AAC encoders add priming samples at the start of a file and padding at the end, because the encoder works in blocks and a recording rarely ends on a block boundary. The M4A records how many samples to skip, so a player trims them and you never hear them.

That instruction has no home in a WAV, so the padding is written out as audio — typically a few dozen milliseconds of near-silence at each end. It is invisible in ordinary listening and it matters in two places: a sample that must trigger instantly, and material that has to line up against picture or against another take. Both are fixed by trimming to the first transient in the editor, which is a ten-second job once you know to look for it.

Why this conversion needs no encoder at all

Every other audio target on this site needs an encoder, and two of them need a WebAssembly package downloaded before they can run. WAV needs neither. Your device already contains an AAC decoder — it is the one that plays the M4A when you double-click it — and a WAV is a RIFF header wrapped around the samples that come out of it, which is arithmetic rather than compression.

So this is the fastest conversion in the audio set and the one with the smallest footprint, and the file never leaves the tab. That is worth something concrete when the material is a client's unreleased stems, a rough vocal or a recording made under an agreement about where it may be stored. Open the network tab and convert something: you will see the page and the advertising that funds it, and nothing carrying your audio.

How to convert an M4A into a WAV file

  1. Drop your M4A onto this page, or click to choose one.
  2. It is decoded with your own device's AAC decoder and written as 16-bit PCM.
  3. Download the WAV and import it into your DAW, sampler or editor.

M4A and WAV compared: a compressed delivery file against a working file

M4A compared with WAV
M4AWAV
Full nameMPEG-4 AudioWaveform Audio
File extension.m4a.wav, .wave
Media typeaudio/mp4audio/wav
CompressionLossy — file size is bought with qualityUncompressed
First published20041991
Published byAppleMicrosoft
SpecificationRIFF WAVE
LicensingPublished, not standardisedPublished, not standardised
Standing todayCurrentCurrent
Bit depth32
Audio channelsup to 48up to 65,535
Opens in a browserEvery browserEvery browser
Considered insteadMP3, FLACFLAC, AIFF

What survives

document properties can cross over — both M4A and WAV have somewhere to store it.

What the target format adds

WAV is a working format and M4A is a finished one. What comes back is editable text and objects rather than a picture of a page, which is usually the reason for the conversion and also where its limits are.

Opening the result

WAV is a container rather than a single format. What actually plays is the codec inside it — usually PCM — which is why two files with the same extension can behave differently on the same device.

The usual programs do not overlap: M4A opens in iTunes, VLC and FFmpeg, WAV in Audacity, Adobe Audition and Reaper — so whoever receives the result needs something from the second list.

What each format is for

The two are aimed at different work: M4A at handing a finished file over and phones, WAV at editing and archiving. That is worth weighing before converting, because the reason one exists is usually the reason the other is awkward.

WAV comes from Microsoft and dates from 1991, specified as RIFF WAVE. Audacity, Adobe Audition and Reaper all read it.

M4A to WAV: what producers and editors ask

Will the WAV sound better than the M4A?

Not if the M4A holds AAC, which is the usual case. AAC permanently discarded parts of the recording and the WAV stores exactly what survived, sample for sample, at roughly five to ten times the size. It sounds identical because it is identical. The exception is real and worth checking: if the M4A holds Apple Lossless, the WAV is a bit-exact copy of an untouched master and nothing at all has been lost.

What sample rate and bit depth do I get?

The sample rate of the source — 44,100 Hz for music from a CD rip or an iTunes purchase, 48,000 Hz for audio taken from video — written at 16 bits per sample in the channel layout the original had. That is CD quality, and it is what every DAW, sampler and hardware player expects to receive. There is no setting to raise it, because raising it would enlarge the file without adding information.

Why does my DAW refuse M4A files?

Usually licensing and dependency management rather than capability. Decoding AAC means either linking a decoder or calling the operating system, and a DAW that must behave identically on Windows and macOS often decides not to. Some hosts will import an M4A and silently convert it to a WAV in the project folder anyway, which is this conversion happening without your involvement. Samplers and hardware are simpler still: they read PCM off a card and have no decoder at all.

How large will the WAV be?

About 10.1 MB per minute for 16-bit stereo at 44,100 Hz — a four-minute track is roughly 40 MB, an hour-long recording is around 600 MB. There is no quality setting because WAV does not compress; that is the whole point of it. A typical AAC M4A of the same track is four to eight megabytes, so expect a five- to tenfold increase.

What happens to the cover art and the tags?

Most of it is lost, and this is where WAV genuinely is a step down. WAV stores metadata in a RIFF INFO chunk, which has room for title, artist, album, track number, genre, date and comment and nothing else — album artist, disc number, description and lyrics have no home there and are dropped, and so is embedded cover art. The chunk is also limited to ISO 8859-1, so a title with Japanese or Cyrillic characters in it is dropped rather than mangled. For a working file that is fine; for archiving a library it is not, and FLAC is the format that keeps everything.

Does the M4A get uploaded?

No. Your device decodes the AAC with the same decoder that plays the file when you double-click it, and the WAV is written by assembling a RIFF header around the samples — no encoder, no extension package, no server. Unreleased material, a client's stems and a rough vocal take all stay on the machine they are on, which you can verify in the network tab while a conversion runs.

More about these formats