Convert OGG to AU — Free Online Tool
Convert OGG audio files (Vorbis or Opus encoded) to Sun AU format using PCM signed 16-bit big-endian encoding — a lossless uncompressed output ideal for Unix/Linux audio toolchains and legacy compatibility. The conversion decodes your compressed OGG stream and writes raw PCM samples into AU's simple binary container.
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FFmpeg Command
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DownloadHow It Works
OGG files typically contain compressed audio streams — most commonly Vorbis or Opus — which use perceptual lossy compression to reduce file size. Converting to AU requires fully decoding that compressed stream back to raw PCM samples, since AU's most common and default codec (pcm_s16be) is uncompressed. FFmpeg reads the OGG container, decodes the Vorbis or Opus audio to raw audio samples in memory, then re-encodes those samples as signed 16-bit big-endian PCM and writes them into the AU container with a minimal fixed-length header. Because the source was lossy-compressed, the output AU file will be lossless PCM but will not recover audio quality that was lost during the original OGG encoding — it reflects the decoded audio faithfully, not the original pre-compression source. Expect a significant increase in file size, as uncompressed 16-bit stereo audio at 44.1 kHz occupies roughly 10 MB per minute compared to a few MB for a typical OGG file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool, which handles decoding the OGG container and its compressed audio stream, then re-encoding to the AU format. |
-i input.ogg
|
Specifies the input OGG file. FFmpeg will detect the container format and identify the audio stream codec — typically Vorbis, Opus, or FLAC — and begin decoding it for conversion. |
-c:a pcm_s16be
|
Sets the output audio codec to signed 16-bit big-endian PCM, which is the standard and default encoding for Sun AU files. This fully decodes the compressed OGG audio (Vorbis or Opus) into raw, uncompressed PCM samples stored in big-endian byte order as required by the AU format's Unix heritage. |
output.au
|
Defines 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-length binary header. |
Common Use Cases
- Feeding decoded audio into Unix command-line audio processing pipelines (e.g., SoX, Audacity batch scripts) that expect raw PCM in AU format
- Preparing audio assets for legacy Sun Solaris or early Java applet environments that natively support AU but not Ogg Vorbis
- Providing uncompressed audio for acoustic analysis or signal processing tools that require byte-accurate PCM samples without lossy codec artifacts
- Converting an OGG podcast or spoken-word recording to AU so it can be imported into older Unix-based audio editing systems without plugin dependencies
- Creating AU files for Java's AudioSystem API, which has built-in support for AU/PCM audio and is commonly used in cross-platform desktop applications
- Archiving a decoded, uncompressed snapshot of an OGG file's audio content for quality inspection or forensic comparison before further processing
Frequently Asked Questions
No — the conversion to uncompressed PCM in AU format does not recover any quality lost during the original OGG encoding. Vorbis and Opus are lossy codecs, meaning some audio information is discarded permanently during compression. The AU file will contain a lossless, byte-accurate representation of the decoded OGG audio, but it reflects the already-compressed signal, not the original studio or source recording. The main benefit of the AU output is the absence of further encoding artifacts, not a restoration of lost data.
OGG with Vorbis or Opus typically achieves 5:1 to 10:1 compression ratios compared to uncompressed audio. The default AU codec (pcm_s16be) stores every audio sample as a raw 16-bit integer with no compression at all. For example, a 3-minute stereo OGG file at 192 kbps might be around 4 MB, while the equivalent AU file at 44.1 kHz 16-bit stereo would be approximately 30 MB. This size increase is expected and unavoidable given the format difference.
No. The Sun AU format uses an extremely minimal fixed-length header that stores only technical parameters: sample rate, channel count, encoding type, and a small annotation field. It has no standardized support for ID3, Vorbis Comment, or any other metadata tag system. Any artist, album, title, or other tags embedded in your OGG file will be silently discarded during conversion to AU.
AU can technically encode a variety of sample rates and supports mono and stereo, but it is a very simple format with limited flexibility compared to modern containers. Most AU consumers expect standard rates like 8000 Hz, 22050 Hz, or 44100 Hz. FFmpeg will carry the OGG file's original sample rate and channel count into the AU output, but if your OGG has an unusual sample rate or more than two channels, compatibility with older AU players or Java's AudioSystem may be reduced. You can add '-ar 44100 -ac 2' to the FFmpeg command to normalize to standard stereo.
Yes. AU supports several other encodings including pcm_s8 (8-bit signed), pcm_u8 (8-bit unsigned), pcm_alaw (G.711 A-law), and pcm_mulaw (G.711 mu-law). To use one, replace '-c:a pcm_s16be' in the command with your preferred codec — for example, 'ffmpeg -i input.ogg -c:a pcm_mulaw output.au' produces a mu-law encoded AU file, which is the traditional 8 kHz telephone-quality format historically used for internet audio clips in the 1990s. Note that 8-bit and G.711 encodings significantly reduce audio fidelity compared to 16-bit PCM.
On Linux or macOS, you can use a shell loop: 'for f in *.ogg; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.ogg}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.ogg) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. Each OGG file in the directory will be decoded and written to a corresponding AU file. This is especially useful for converting large collections of OGG sound effects or samples into AU for use in a Unix-based audio pipeline.
Technical Notes
The Sun AU format (.au) is one of the oldest digital audio file formats, predating WAV and MP3, and its structure is intentionally minimal: a 6-field binary header followed by a flat stream of PCM samples. The default codec for this conversion is pcm_s16be — signed 16-bit samples stored in big-endian byte order, reflecting AU's origins on Sun SPARC and Motorola architectures where big-endian is native. On modern x86/x64 systems (which are little-endian), byte order is handled transparently by FFmpeg. One notable limitation is that AU has no support for chapters, multiple audio tracks, or subtitle streams — features that OGG containers can carry — so any such content in the source file is simply ignored. If the OGG file contains a FLAC audio stream (lossless), the AU output will be a true lossless-to-lossless conversion at the PCM level, with no generational quality loss. If the source is Vorbis or Opus, the conversion is a lossy-to-uncompressed path. The AU format is natively supported by Java's javax.sound.sampled API without additional libraries, making it a practical choice for Java desktop application audio assets.