Convert AU to AAC — Free Online Tool
Convert Sun AU audio files to AAC format using FFmpeg in your browser. This tool decodes PCM audio stored in AU's simple Unix container and re-encodes it with AAC's perceptual compression, producing a modern, compact audio file compatible with iOS, Android, iTunes, and streaming platforms.
to
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
AU files store audio as uncompressed PCM data (typically 16-bit big-endian signed PCM, as AU originated on Sun's SPARC workstations) inside a minimal header structure. Since AAC is a completely different codec — a lossy transform-based codec using modified discrete cosine transform (MDCT) compression — this conversion cannot simply remux the stream; it must fully decode the raw PCM samples from the AU container and then re-encode them using FFmpeg's built-in AAC encoder. The result is a significantly smaller file that sacrifices some audio fidelity in exchange for broad compatibility with modern devices and streaming services. The default 128k bitrate is a sensible balance for most speech and music content originating from AU files.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all decoding, encoding, and format detection for this conversion. In the browser version of this tool, FFmpeg runs via WebAssembly (ffmpeg.wasm) with no server involvement. |
-i input.au
|
Specifies the input AU file. FFmpeg reads the AU header to detect the PCM encoding variant (e.g., pcm_s16be, pcm_mulaw) and sample rate, then demuxes the raw audio data for decoding. |
-c:a aac
|
Selects FFmpeg's built-in AAC encoder to compress the decoded PCM audio. This encoder produces AAC-LC (Low Complexity) output, the AAC profile with the broadest device and platform compatibility, including iOS, Android, and web browsers. |
-b:a 128k
|
Sets the AAC encoder's target bitrate to 128 kilobits per second. This is the standard default that balances file size reduction against audio quality for typical AU source material such as voice recordings and general-purpose audio. Increase to 192k or 256k for music-quality output. |
output.aac
|
Defines the output filename and extension. The .aac extension tells FFmpeg to write a raw AAC bitstream file, which is directly playable by Apple devices, modern browsers, and most media players without any additional container wrapping. |
Common Use Cases
- Modernizing legacy Unix system audio files or sound effects originally distributed in AU format for use on smartphones, tablets, or streaming platforms that don't support AU playback.
- Converting AU audio clips from early Java applets or late-1990s web pages — which commonly used AU as the browser audio format — into AAC for archival or re-publication.
- Preparing Sun workstation voice recordings or system sounds for playback in iOS or Android apps, which natively support AAC but have no AU decoder.
- Compressing large uncompressed AU audio files (PCM can be several times larger than compressed formats) for distribution via email, messaging apps, or web embeds.
- Migrating audio assets from legacy Sun Solaris or early Unix-based multimedia projects into a modern format compatible with iTunes, Apple Music, or podcast hosting platforms.
- Converting AU-format audio from scientific or telecommunications research datasets into AAC for use in presentations, reports, or sharing with collaborators on standard media players.
Frequently Asked Questions
It depends on the content and the target bitrate. AU files typically contain uncompressed PCM audio, meaning they carry the full original signal with no compression artifacts. AAC at 128k is lossy — it permanently discards audio information the codec deems less perceptible. For voice recordings and speech, 128k AAC is generally transparent. For high-fidelity music content, you may want to increase to 192k or 256k using the -b:a flag to minimize audible difference.
AU files store audio as raw PCM (e.g., pcm_s16be — 16-bit signed big-endian samples), while AAC is a transform-based compressed codec. These are fundamentally incompatible audio representations — there is no way to wrap PCM data inside an AAC file without actually compressing it. FFmpeg must fully decode the PCM stream and run it through the AAC encoder, which is a computationally heavier process than a simple remux.
AU files with PCM audio are uncompressed and can be very large — a one-minute stereo file at 16-bit/44.1kHz occupies roughly 10MB. The same audio converted to AAC at 128k would be around 1MB, a reduction of approximately 90%. This dramatic size difference is the primary practical reason to convert AU files to AAC for distribution or web use.
The AU format uses an extremely minimal header that includes only technical fields: a magic number, data offset, data size, encoding type, sample rate, and channel count. It has no standardized fields for ID3-style metadata such as artist, title, or album. As a result, your output AAC file will not contain any embedded metadata tags, since there was nothing to carry over from the source.
Replace the value after -b:a in the command. For example, to encode at 192k instead of the default 128k, use: ffmpeg -i input.au -c:a aac -b:a 192k output.aac. Common choices are 64k for voice/speech, 128k for general use, and 192k–256k for music where quality is a priority. Note that going above 256k provides diminishing returns with the AAC codec.
The command shown processes a single file, but you can adapt it for batch conversion in a shell. On Linux or macOS, run: for f in *.au; do ffmpeg -i "$f" -c:a aac -b:a 128k "${f%.au}.aac"; done. On Windows Command Prompt, use: for %f in (*.au) do ffmpeg -i "%f" -c:a aac -b:a 128k "%~nf.aac". This is especially useful when migrating large collections of legacy AU audio assets.
Yes. AAC is Apple's preferred audio format, and AAC files with the .aac extension are natively supported by iOS, macOS, iTunes, and Apple Music. FFmpeg's built-in AAC encoder produces standards-compliant AAC-LC output, which is the AAC profile Apple devices use for playback. If you require AAC encoded with Apple's own encoder for maximum compatibility with specific iTunes features, you would need to use Apple's tools directly.
Technical Notes
AU (Sun Audio) files most commonly encode audio as pcm_s16be — 16-bit signed big-endian PCM — though the format also supports 8-bit signed, 8-bit unsigned, A-law, and mu-law (G.711) variants. FFmpeg will automatically detect which PCM encoding is in use and decode it correctly regardless of variant. The output AAC file uses FFmpeg's native AAC encoder (codec: aac), which produces AAC-LC (Low Complexity) profile output — the most universally supported AAC variant. An alternative is libfdk_aac, which is generally considered higher quality but is not included in standard FFmpeg builds due to licensing restrictions. AU files carry no stereo/mono metadata beyond a raw channel count, so FFmpeg will preserve the channel configuration faithfully. Since AU has no subtitle, chapter, or multi-track support and neither does a bare .aac file, there is nothing lost on those fronts. One edge case: very old AU files from Sun workstations may use non-standard sample rates (e.g., 8012 Hz for telephone audio); AAC handles these fine, but some playback software may behave unexpectedly with non-standard rates.