Convert FLAC to AU — Free Online Tool
Convert FLAC lossless audio to Sun AU format using PCM 16-bit big-endian encoding — the AU format's default codec and a staple of Unix/Linux audio pipelines. This tool runs entirely in your browser via FFmpeg.wasm, so your audio files never leave your machine.
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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
FLAC stores audio as losslessly compressed integer PCM data. During this conversion, FFmpeg decodes the FLAC stream back to raw PCM samples, then re-encodes them as 16-bit signed big-endian PCM (pcm_s16be) and wraps them in the AU container — a minimal header format developed by Sun Microsystems. Because both FLAC and AU's default codec are lossless PCM representations, no audio quality is lost in the conversion; the decoded samples are numerically identical to what FLAC stored. The only meaningful change is the container and the elimination of FLAC's compression, which means the output AU file will be significantly larger than the input FLAC file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the underlying engine that handles all decoding, sample format conversion, and encoding for this FLAC-to-AU conversion. |
-i input.flac
|
Specifies the input FLAC file. FFmpeg reads the FLAC container, parses its metadata headers, and decodes the losslessly compressed audio stream back into raw PCM samples for processing. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard and most compatible codec for the AU container. This matches the big-endian byte order of AU's Sun Microsystems origins and ensures broad compatibility with Unix audio tools and Java's sound API. |
output.au
|
Specifies the output filename with the .au extension. FFmpeg uses this extension to automatically select the Sun AU container format, which wraps the pcm_s16be audio stream in AU's minimal fixed header structure. |
Common Use Cases
- Feeding audio into legacy Unix or Solaris audio tools that only accept the AU (.au or .snd) format natively
- Preparing audio samples for Java applications using the javax.sound API, which has built-in support for AU/PCM files without additional libraries
- Converting archival FLAC recordings into a flat, uncompressed PCM format for ingest into older digital audio workstations or hardware samplers that do not support compressed input
- Generating AU files for use in early web audio contexts or emulators that replicate 1990s Unix workstation environments
- Providing uncompressed raw PCM audio in a simple, self-describing container for signal processing scripts that parse AU headers to extract sample rate and bit depth metadata
- Creating test audio files in AU format for audio codec development and testing pipelines running on Unix/Linux servers
Frequently Asked Questions
No — this specific conversion is fully lossless. FLAC is a lossless format, and the AU file is encoded as pcm_s16be (16-bit signed big-endian PCM), which is also lossless. As long as your FLAC source is 16-bit, the decoded samples written into the AU file are numerically identical to the originals. If your FLAC file is 24-bit, FFmpeg will dither it down to 16-bit for the AU container, which involves a minor precision reduction.
FLAC uses lossless compression to shrink audio data, typically achieving 40–60% size reduction compared to raw PCM. The AU format with pcm_s16be codec stores audio as completely uncompressed PCM samples with no compression whatsoever, so the output file size reflects the true raw size of the audio data — roughly 10 MB per minute for stereo 44.1kHz 16-bit audio. This is expected behavior and not a sign that anything went wrong.
No. The Sun AU format uses an extremely minimal header that stores only essential technical parameters: the byte offset to audio data, data size, encoding type, sample rate, and channel count. It has no standardized support for metadata fields like artist, album, title, or track number. All Vorbis comment tags embedded in your FLAC file will be discarded during conversion to AU.
Yes — the AU container supports several other codecs including pcm_s8 (8-bit signed PCM), pcm_u8 (8-bit unsigned PCM), pcm_alaw (G.711 A-law, lossy), and pcm_mulaw (G.711 µ-law, lossy). To change the codec, modify the FFmpeg command by replacing 'pcm_s16be' with your desired codec, for example: ffmpeg -i input.flac -c:a pcm_mulaw output.au. Note that A-law and µ-law are lossy and are typically used for telephony applications.
You can use a shell loop to batch process files. On Linux or macOS, run: for f in *.flac; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.flac}.au"; done. On Windows PowerShell, use: Get-ChildItem *.flac | ForEach-Object { ffmpeg -i $_.FullName -c:a pcm_s16be ($_.BaseName + '.au') }. The browser-based tool processes one file at a time, but the FFmpeg command shown works on your desktop for any number of files.
The AU format can technically encode multiple channels and arbitrary sample rates, as these values are stored in its header. However, AU is a very old format with limited software support beyond stereo configurations, and many AU-compatible applications assume mono or stereo input. High sample rates like 96kHz or 192kHz from hi-res FLAC files will be stored correctly in the AU header, but playback compatibility in legacy software is not guaranteed.
Technical Notes
The Sun AU format (.au) is one of the oldest digital audio container formats still in use, featuring a fixed or variable-length header followed by raw audio sample data. Its simplicity — just six mandatory header fields — makes it trivial to parse programmatically, which is why it remains relevant in Java (javax.sound.sampled), scientific computing, and Unix legacy contexts. The default codec chosen for this conversion is pcm_s16be (signed 16-bit big-endian PCM), which matches the AU format's historical roots on big-endian Sun SPARC hardware. Unlike WAV, which uses little-endian PCM (pcm_s16le) to match x86 architecture, AU's big-endian byte order may require explicit handling in software written for modern little-endian processors. FLAC's replay gain tags, cue sheet data, and seek table are all stripped during conversion, as the AU container has no equivalent structures. If your FLAC source contains audio at bit depths above 16-bit, be aware that pcm_s16be truncates to 16 bits; for 24-bit FLAC sources, consider using pcm_s8 for 8-bit output or staying within the AU codec set. The output AU file has no internal compression, so storage and streaming bandwidth costs are higher than the source FLAC.