Convert M4A to AU — Free Online Tool
Convert M4A files (AAC-encoded audio from iTunes, podcasts, or Apple devices) to AU format, the classic Unix audio container developed by Sun Microsystems. The conversion decodes AAC audio and re-encodes it as uncompressed PCM signed 16-bit big-endian audio — producing a raw, lossless-quality AU file compatible with legacy Unix tools and Java Sound API applications.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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 store audio compressed with the AAC codec inside an MPEG-4 container. AU is a fundamentally different format: it uses a minimal header followed by raw PCM audio data, with no support for modern compressed codecs like AAC. This means the conversion cannot simply remux the stream — the AAC audio must be fully decoded back to raw PCM samples and then written as signed 16-bit big-endian PCM (pcm_s16be) into the AU container. The result is uncompressed audio that is essentially lossless in terms of what was audible in the AAC file, though any quality lost during the original AAC encoding cannot be recovered. File sizes will increase substantially compared to the compressed M4A source. iTunes metadata, chapter markers, and gapless playback tags supported by M4A are not carried over, as the AU format has no mechanism to store them.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In the browser-based version of this tool, FFmpeg runs entirely via WebAssembly (FFmpeg.wasm) with no server involvement — the same command shown here can be run locally on your desktop with a standard FFmpeg installation. |
-i input.m4a
|
Specifies the input file — an M4A container holding AAC-encoded audio, typically sourced from iTunes purchases, Apple Voice Memos, or podcast downloads. FFmpeg reads the MPEG-4 container and demuxes the AAC audio stream for decoding. |
-c:a pcm_s16be
|
Instructs FFmpeg to encode the output audio as PCM signed 16-bit big-endian, the standard uncompressed audio encoding for AU files. This decodes the AAC audio from the M4A and writes raw 16-bit samples in big-endian byte order — the native format for Sun Microsystems AU files — rather than applying any further compression. |
output.au
|
Defines the output filename with the .au extension, which tells FFmpeg to write a Sun AU container. The AU container wraps the pcm_s16be audio data with a minimal header containing the sample rate, channel count, and encoding type, producing a file immediately usable by Unix audio tools and Java Sound API applications. |
Common Use Cases
- Preparing audio samples for use with Java applications that use the Java Sound API, which has native support for AU/SND files
- Supplying audio to legacy Unix or Solaris workstation software that expects .au files as its native input format
- Converting podcast or audiobook M4A files into a simple, headerless-style PCM format for ingestion into DSP research pipelines or signal processing tools that require uncompressed audio
- Creating AU audio assets for embedding in older web pages or Java applets that referenced .au files as their audio source in the early internet era
- Stripping AAC compression from an M4A recording to obtain raw PCM audio data in a portable, universally readable binary format for archival or analysis
Frequently Asked Questions
The conversion from M4A to AU will not introduce any new lossy compression — the AAC stream is decoded to raw PCM and written uncompressed into the AU file. That said, this is not a true lossless recovery: any quality lost when the audio was originally encoded as AAC cannot be restored. The output AU file faithfully represents the decoded audio from the M4A, no better and no worse.
M4A files use AAC compression, which typically reduces audio file size by a factor of 5–10x compared to uncompressed audio. AU files store audio as raw uncompressed PCM data, so a 5-minute M4A encoded at 128k bitrate (around 5MB) will expand to roughly 50MB as a 16-bit stereo AU file at 44.1kHz. This size increase is expected and unavoidable given the fundamental difference between compressed and uncompressed audio storage.
No. The AU format uses an extremely minimal header that stores only sample rate, encoding type, and channel count — it has no concept of ID3 tags, iTunes metadata fields, chapter markers, album art, or gapless playback information. All of this structured metadata present in the M4A will be lost in the conversion. If preserving metadata is important, consider converting to a format like FLAC or WAV instead.
pcm_s16be stands for PCM signed 16-bit big-endian — it describes uncompressed audio samples stored as 16-bit signed integers in big-endian byte order (most significant byte first). The AU format was developed by Sun Microsystems on SPARC-based Unix systems, which used big-endian architecture, so big-endian byte ordering is the native and most widely supported encoding for AU files. This is in contrast to WAV files, which use little-endian PCM. FFmpeg defaults to pcm_s16be when targeting AU output for maximum compatibility with Unix tools and Java Sound.
Yes. The AU format supports several audio encodings beyond pcm_s16be, including pcm_s8, pcm_u8, pcm_alaw (G.711 A-law), and pcm_mulaw (G.711 mu-law). You can change the codec by replacing -c:a pcm_s16be in the FFmpeg command with your preferred codec, for example: ffmpeg -i input.m4a -c:a pcm_mulaw output.au. The mu-law and A-law variants are lossy compressed and commonly used for telephony audio, while pcm_s8 reduces bit depth to 8 bits for smaller but lower-quality output.
You can adapt the displayed FFmpeg command into a shell loop for batch processing. On Linux or macOS, run: for f in *.m4a; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4a}.au"; done. On Windows Command Prompt, use: for %f in (*.m4a) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au". This processes every M4A file in the current directory and outputs a corresponding AU file with the same base name. Batch processing in the browser tool requires converting files one at a time.
Technical Notes
The AU format's simplicity is both its strength and its limitation for this conversion. Its header contains only four fields — magic number, data offset, data size, encoding type, sample rate, and channel count — which means there is virtually no metadata overhead, but also no room for any of the rich tagging that M4A/iTunes supports. The default output codec pcm_s16be delivers a dynamic range of 96dB (standard CD quality at 16 bits), which is sufficient to faithfully represent typical AAC-encoded music or podcast audio. If the source M4A was encoded at a high sample rate (e.g., 48kHz), FFmpeg will preserve that sample rate in the AU output rather than resampling, so the output quality is directly tied to the source. One important compatibility note: some older Unix tools and Java Sound API implementations expect AU files with specific sample rates (8kHz for telephony or 44.1kHz for music); ensure your downstream tool supports the sample rate of your source M4A. The AU format does not support multi-channel audio beyond stereo in many implementations, so surround-encoded M4A files may require additional downmixing flags. Finally, because AU stores raw PCM, the format is well-suited for streaming or piping audio data between Unix processes, which was its original design purpose.