Convert AU to OGG — Free Online Tool
Convert Sun AU audio files to OGG Vorbis format directly in your browser — no uploads required. This conversion re-encodes raw PCM audio (typically 16-bit big-endian) from AU's uncompressed Unix legacy format into Vorbis, a modern lossy codec that dramatically reduces file size while maintaining excellent perceptual quality.
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 (most commonly 16-bit big-endian signed PCM, or occasionally 8-bit, A-law, or μ-law variants) inside a minimal header structure originally designed for Sun workstations. During this conversion, FFmpeg decodes the raw PCM stream from the AU container and re-encodes it using the libvorbis encoder, wrapping the result in the OGG container format developed by Xiph.Org. Because AU is uncompressed and OGG Vorbis is a lossy format, this is a full transcode — not a remux — meaning the audio signal is decoded to PCM and then compressed using Vorbis's psychoacoustic model. The default quality level of -q:a 4 produces variable bitrate output roughly equivalent to 128–160 kbps, which is transparent for most listeners while shrinking the file to a fraction of its original AU size.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool, which handles decoding the Sun AU input, transcoding the audio, and writing the OGG Vorbis output. |
-i input.au
|
Specifies the input Sun AU file. FFmpeg reads the AU header to determine the sample rate, channel count, and PCM encoding variant (such as pcm_s16be, pcm_alaw, or pcm_mulaw) before decoding the audio stream. |
-c:a libvorbis
|
Selects the libvorbis encoder to compress the decoded PCM audio using the Vorbis codec — a free, open, and patent-unencumbered lossy audio format that is the default audio codec for the OGG container. |
-q:a 4
|
Sets the Vorbis variable bitrate quality to level 4 on a 0–10 scale, targeting approximately 128–160 kbps output. This is the standard balanced setting that produces transparent audio for most content while reducing the uncompressed AU file size by roughly 8–10x. |
output.ogg
|
Defines the output filename and tells FFmpeg to write the encoded Vorbis audio stream into an OGG container, the open format maintained by Xiph.Org that is widely supported in browsers, Linux audio systems, and open-source media players. |
Common Use Cases
- Publishing legacy Unix system audio samples or sound effects (originally distributed as .au files) to web platforms that support OGG Vorbis for open-format audio playback
- Compressing old Sun workstation or NeXT-era audio archives — where AU files can be tens of megabytes of raw PCM — into manageable OGG files for archival or distribution
- Converting AU audio assets used in early Java applets or Unix applications into OGG for use in modern HTML5 games or web apps that prefer open royalty-free formats
- Preparing AU-format telephone recordings (A-law or μ-law encoded) for editing or playback in audio software that handles OGG Vorbis more natively than raw AU
- Reducing bandwidth and storage costs when serving large libraries of .au sound clips from a web server by re-encoding them as compressed OGG Vorbis files
- Migrating audio content from legacy Unix workstations or Solaris systems to cross-platform open-format audio pipelines that standardize on OGG
Frequently Asked Questions
Yes — OGG Vorbis is a lossy codec, so converting from AU's uncompressed PCM introduces some degree of audio degradation. However, at the default quality setting (-q:a 4, approximately 128–160 kbps VBR), the loss is psychoacoustically masked and inaudible to most listeners on typical content. If you are archiving source material and quality is paramount, consider encoding with FLAC inside OGG instead, which is lossless and supported by the same OGG container.
Yes. FFmpeg fully supports the A-law (pcm_alaw) and μ-law (pcm_mulaw) codecs that AU files commonly use for telephony audio. During conversion, FFmpeg automatically decodes whichever PCM variant is present in the AU file to an intermediate linear PCM representation, then re-encodes that to OGG Vorbis. You do not need to do anything special — the same ffmpeg command works regardless of which AU codec variant your file uses.
Adjust the -q:a value, which controls Vorbis's variable bitrate quality scale. The scale runs from 0 (lowest quality, smallest file, ~64 kbps) to 10 (highest quality, largest file, ~500 kbps). The default of 4 is a solid general-purpose setting. For example, use -q:a 6 for higher fidelity (around 192 kbps) or -q:a 2 for aggressive compression (around 96 kbps). Substitute your chosen value directly into the command: ffmpeg -i input.au -c:a libvorbis -q:a 6 output.ogg.
The AU format has virtually no metadata support — its header contains only sample rate, encoding type, and a minimal annotation field. OGG Vorbis, by contrast, uses Vorbis Comment tags and supports rich metadata including artist, album, title, genre, and track number. After conversion, you can add or edit these tags using tools like Kid3 or MusicBrainz Picard, which is one practical advantage of migrating from AU to OGG.
The browser tool processes one file at a time, but on your desktop you can batch convert using a shell loop. On Linux or macOS, run: for f in *.au; do ffmpeg -i "$f" -c:a libvorbis -q:a 4 "${f%.au}.ogg"; done. On Windows Command Prompt, use: for %f in (*.au) do ffmpeg -i "%f" -c:a libvorbis -q:a 4 "%~nf.ogg". This applies the same conversion settings to every AU file in the current directory.
AU files store audio as uncompressed PCM, so a one-minute stereo AU file at 44.1 kHz 16-bit occupies roughly 10 MB. At the default -q:a 4 Vorbis setting (approximately 128–160 kbps), the equivalent OGG file will be around 1–1.2 MB — a compression ratio of roughly 8–10x. For AU files using 8-bit or telephone-bandwidth μ-law/A-law encoding, the original files are already smaller, but OGG Vorbis will still typically achieve 3–5x further reduction.
Technical Notes
AU files originate from Sun Microsystems and use a big-endian byte order for their PCM samples, which is the opposite of the little-endian convention used on most modern x86 systems. FFmpeg handles this byte-swapping transparently during decoding. The default AU codec (pcm_s16be) delivers CD-quality audio resolution, so no upsampling or downsampling occurs unless the source AU file uses an unusual sample rate. OGG Vorbis does not support arbitrary sample rates as cleanly as raw PCM — very unusual rates may be rounded to the nearest standard rate during encoding. The OGG container used here supports multiple audio tracks and chapter markers, though a single-stream AU file will produce a single-stream OGG. One important limitation: Vorbis encoding introduces a small amount of encoder latency (pre-roll/priming frames), which means the OGG file's playback start is slightly padded; this is standard Vorbis behavior and not a bug. If lossless output is required, replacing -c:a libvorbis -q:a 4 with -c:a flac will encode the audio as FLAC inside the OGG container, preserving every sample from the original AU file exactly.