Convert MP3 to AU — Free Online Tool
Convert MP3 files to Sun AU format using PCM signed 16-bit big-endian audio, decoded and re-encoded entirely in your browser. This tool is ideal for Unix/Linux workflows and legacy systems that expect the simple .au container with uncompressed PCM audio rather than lossy compressed streams.
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FFmpeg Command
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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
MP3 uses lossy compression via the MPEG Audio Layer III codec (libmp3lame), storing audio as perceptually encoded data rather than raw samples. Converting to AU requires fully decoding the MP3 bitstream back into raw PCM samples, then writing those samples into the Sun AU container using the pcm_s16be codec — 16-bit signed integers in big-endian byte order. This is a decode-then-re-encode operation, not a remux: every compressed MP3 frame is decompressed, and the resulting audio is stored uncompressed. Because AU with pcm_s16be is lossless at the PCM level, no additional quality is lost beyond what was already lost when the original MP3 was created. The resulting .au file will be significantly larger than the source MP3 because uncompressed PCM requires roughly 10x more storage than typical MP3 bitrates.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, the underlying engine that also powers the in-browser conversion via FFmpeg.wasm compiled to WebAssembly. |
-i input.mp3
|
Specifies the input file as an MP3, telling FFmpeg to open and demux the MPEG Audio Layer III bitstream in preparation for decoding. |
-c:a pcm_s16be
|
Sets the audio codec for the output to PCM signed 16-bit big-endian — the uncompressed format native to the Sun AU container. This fully decodes the lossy MP3 frames into raw audio samples and re-encodes them as big-endian integers, which is what AU-reading Unix applications expect. |
output.au
|
Specifies the output filename with the .au extension, which causes FFmpeg to mux the PCM audio stream into the Sun AU container format with its characteristic minimal binary header. |
Common Use Cases
- Providing audio samples to Unix or Solaris-based systems or legacy software pipelines that natively read .au files but cannot decode MP3
- Preparing audio for use with older Java applications or applets that relied on Sun's AudioSystem API, which has historically favored the .au format with PCM or mulaw encoding
- Converting MP3 music or sound effects into a headerless-friendly PCM container for signal processing tools running on BSD or early Unix environments
- Supplying uncompressed audio to scientific or audio analysis software that requires raw PCM samples in a simple container without complex codec negotiation
- Archiving or testing audio routing on systems where .au is the default playback format (such as early NeXTSTEP or Solaris audio subsystems)
- Stripping MP3 compression artifacts concern aside and producing a flat PCM representation of an MP3 recording for waveform inspection or manual editing in hex-based tools
Frequently Asked Questions
No — the quality ceiling is permanently set by the original MP3 encoding. The MP3 format is lossy, meaning frequency data discarded during compression cannot be recovered. Converting to AU with pcm_s16be produces an uncompressed file, but it is an uncompressed representation of already-degraded audio. Think of it as a lossless copy of a lossy file: no further quality is lost, but no quality is restored either.
MP3 achieves compression ratios of roughly 10:1 or better by discarding perceptually redundant audio data. AU with pcm_s16be stores every audio sample as a raw 16-bit integer with no compression, so a 3-minute MP3 at 128k (roughly 2.8 MB) will expand to approximately 30 MB as a 44.1 kHz stereo PCM AU file. This is expected and unavoidable when converting from a compressed to an uncompressed format.
The AU container supports several codecs including pcm_s16be, pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw. The tool defaults to pcm_s16be (16-bit signed big-endian PCM) because it offers the best dynamic range and fidelity among the available options, making it the most suitable for general-purpose audio derived from a 128k+ MP3 source. If your target system requires mulaw (common in telephony or early Sun workstations), you would need to modify the FFmpeg command manually.
No. The Sun AU format has an extremely minimal header — it stores only essential technical metadata like sample rate, channel count, and encoding type. It has no facility for ID3 tags or any equivalent tagging system. All artist, title, album, and cover art metadata embedded in your MP3 will be discarded during conversion. If preserving metadata is important, keep your original MP3 or export to a format like FLAC or OGG Vorbis instead.
Replace '-c:a pcm_s16be' with '-c:a pcm_mulaw' in the command to encode using G.711 mu-law, which was the default encoding on early Sun workstations and many telephony systems. For 8-bit PCM you can use '-c:a pcm_s8' or '-c:a pcm_u8'. Note that mulaw and alaw are lossy companding codecs optimized for voice, so they are not ideal for music. Example: 'ffmpeg -i input.mp3 -c:a pcm_mulaw output.au'.
Yes. On Linux or macOS you can use a shell loop: 'for f in *.mp3; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mp3}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.mp3) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. This will process each MP3 in the current directory and produce a corresponding .au file with the same base filename.
Technical Notes
The Sun AU format (.au, also known as SND) uses a simple 24-byte minimum header containing a magic number, data offset, data size, encoding type, sample rate, and channel count — making it one of the most straightforward audio container formats in existence. When FFmpeg converts MP3 to AU using pcm_s16be, it writes 16-bit big-endian samples, which means byte order matters: systems with little-endian architectures (x86, ARM) must byte-swap during playback, which all modern AU-capable decoders handle transparently. The sample rate and channel count from the source MP3 are preserved by default — a 44.1 kHz stereo MP3 produces a 44.1 kHz stereo AU file. One important limitation is that AU has no support for seeking by time in streams without a known data size, and FFmpeg may write 0xFFFFFFFF as the data size field when the output size is not known in advance (e.g., during piped output), though writing directly to a named file avoids this. No chapters, subtitle streams, or multiple audio tracks are supported by either format in this conversion.