Convert M4A to WAV — Free Online Tool
Convert M4A files (AAC-encoded audio from iTunes, Apple Music, or podcasts) to WAV format using lossless PCM encoding in your browser. The AAC audio stream is fully decoded and written as uncompressed 16-bit PCM — ideal for audio editing, broadcast preparation, or archiving without any further generation loss.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your M4A file here
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Settings
Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
M4A files typically contain AAC-encoded audio, a lossy compressed format that achieves small file sizes by discarding audio data the human ear is unlikely to notice. During conversion to WAV, FFmpeg fully decodes the AAC bitstream back to raw PCM samples and writes them into a WAV container using the pcm_s16le codec — 16-bit signed little-endian PCM. This is a decode-then-rewrite operation, not a remux: the compressed AAC data cannot be placed inside a WAV container, so full decoding is mandatory. The resulting WAV file is uncompressed and will be significantly larger than the source M4A. Importantly, because AAC is a lossy format, the WAV output reflects the audio quality of the original AAC encode — decoding to PCM does not recover information discarded during the original AAC compression, so this is not a true lossless chain from the original recording unless the M4A source was itself encoded from a lossless master.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version of this tool, FFmpeg runs entirely via WebAssembly (FFmpeg.wasm) with no server involvement — this command is also shown so you can run the identical conversion locally on your machine. |
-i input.m4a
|
Specifies the input file — your M4A source containing AAC-encoded audio in an MPEG-4 container. FFmpeg reads the container, identifies the AAC audio stream, and prepares it for decoding. |
-c:a pcm_s16le
|
Sets the audio codec for the output to pcm_s16le — 16-bit signed little-endian PCM. This fully decodes the compressed AAC audio from the M4A and writes it as raw, uncompressed samples into the WAV container, which is the industry-standard format for uncompressed audio interchange. |
output.wav
|
Specifies the output filename with a .wav extension. FFmpeg uses this extension to determine that the output container should be WAV (RIFF Waveform Audio), which pairs correctly with the pcm_s16le codec to produce a standard uncompressed audio file. |
Common Use Cases
- Importing iTunes or Apple Music purchases into a DAW (like Pro Tools, Logic Pro, or Ableton) that requires uncompressed WAV files for editing or mixing sessions.
- Preparing podcast episode masters originally recorded or distributed as M4A for submission to broadcast radio stations that mandate WAV or AIFF deliverables.
- Using AAC-encoded audiobook chapters stored in M4A as source material for audio post-production where frame-accurate editing demands uncompressed PCM.
- Converting M4A ringtones or sound effects downloaded from Apple platforms into WAV format for use in game engines like Unity or Unreal Engine that expect PCM audio assets.
- Archiving an M4A music library to WAV so that future format conversions start from an uncompressed intermediate, avoiding additional lossy re-encoding artifacts.
- Supplying WAV files to hardware samplers, drum machines, or DJ controllers that do not support AAC or M4A playback natively.
Frequently Asked Questions
No — decoding AAC to PCM WAV does not recover any audio quality lost during the original AAC compression. The WAV file will be an uncompressed, bit-for-bit accurate representation of the decoded AAC signal, but any frequencies or details discarded when the M4A was first encoded are gone permanently. What you gain is a format that introduces no further compression artifacts in subsequent processing or re-encoding steps.
AAC compression inside M4A typically achieves 7:1 to 15:1 size reduction compared to uncompressed audio. The pcm_s16le codec used in the WAV output stores every sample as a raw 16-bit integer with no compression whatsoever. A 5-minute M4A at 128kbps (~5MB) will expand to roughly 50MB as a 16-bit stereo WAV at 44.1kHz — this is expected and normal behavior, not an error.
Mostly no. M4A files use iTunes-style metadata atoms (such as ©nam, ©ART, and covr for album art) that are not natively supported by the WAV container format. The WAV specification includes only a limited INFO chunk for basic text tags, and album art is not supported at all. Some players can read ID3 tags embedded in WAV files, but FFmpeg's default conversion does not write these, so you should expect most metadata to be lost in the output WAV.
No. WAV does not support chapters as a container feature, so any chapter markers present in an M4A file (common in audiobooks and long-form podcast files) will be silently discarded during conversion. If you need to preserve chapters, consider splitting the M4A into individual chapter files before converting, using a tool like mp4chaps or ffmpeg with chapter-based segment splitting.
Replace pcm_s16le with a different PCM codec to change the bit depth. Use pcm_s24le for 24-bit output (preferred for professional audio work), pcm_s32le for 32-bit integer, or pcm_f32le for 32-bit float. For example: ffmpeg -i input.m4a -c:a pcm_s24le output.wav. Higher bit depths produce larger files but offer more headroom for editing and processing without clipping or quantization noise.
Yes, with a simple shell loop. On Linux or macOS: for f in *.m4a; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.m4a}.wav"; done. On Windows Command Prompt: for %f in (*.m4a) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This applies the same pcm_s16le conversion to every M4A in the current directory and outputs a matching WAV file for each.
Technical Notes
The conversion from M4A to WAV is a mandatory full decode operation because WAV does not support AAC or any MPEG compressed audio codec in any widely recognized form — the pcm_s16le codec used here produces standard 16-bit little-endian PCM, which is the WAV format's canonical uncompressed representation and is readable by virtually every piece of audio software ever written. M4A files can technically contain audio encoded with codecs other than AAC — including Apple Lossless (ALAC) — and if your source M4A contains ALAC, the conversion to 16-bit PCM WAV will be a true lossless transcode (subject only to any bit-depth reduction if the ALAC source exceeds 16 bits). Gapless playback metadata present in the M4A (iTunSMPB tags that define encoder delay and padding) is not propagated to WAV, which could affect seamless playback of album-length audio if the WAV is later re-encoded. The WAV container has a theoretical 4GB file size limit due to its 32-bit chunk size header — for very long high-sample-rate audio, the RF64/W64 extension would be needed, though this is rarely a concern for typical music or podcast files. Sample rate and channel count are preserved as-is from the M4A source.