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Supports: OGG
.ogg is accepted on this page; for .oga containers use OGA to AU. Batch upload is supported./dev/audio legacy), 22050 Hz (simple FX, AM-radio quality), 44100 Hz (CD quality, the safest default for Java javax.sound.sampled playback), or 48000 Hz (broadcast / DAW projects).HH:MM:SS.sss to crop a region — handy for keeping output files small, since AU is uncompressed. Click Convert and download the .au file. Conversion runs on our servers — no sign-up, no watermark.OGG (Ogg Vorbis) is a free, royalty-free lossy container from the Xiph.Org Foundation, popular in games (the Source engine, Unity, Godot all ship Vorbis decoders) and for podcast/streaming pipelines. AU — magic bytes .snd (0x2e736e64), big-endian header — was introduced by Sun Microsystems in the early 1990s and remains the de facto Unix audio format. Converting OGG into AU is uncommon as an end-user task but very common in development and research workflows.
javax.sound.sampled playback — AudioFileFormat.Type.AU is one of only five formats the standard Java Sound API recognizes natively (alongside WAVE, AIFF, AIFC, SND). Shipping .au assets means you can AudioSystem.getAudioInputStream(...) without bundling JOrbis or a Vorbis SPI./dev/audio device, so init scripts, alarm chimes, and SPARCstation-era applications still expect .au. NetBSD and OpenBSD audio tooling still handle AU natively.audioread() accepts .au directly; many older signal-processing datasets (NIST TIMIT, CMU ARCTIC supplements, telephony corpora) are distributed as 8 kHz mu-law AU.sox, play, ffmpeg, and Audacity all read/write AU without plugins, so AU is a portable interchange format when you want a deterministic, header-defined PCM file for build pipelines..au file with about 30 lines of C — far simpler than parsing Vorbis bitstream packets.| Property | OGG (Vorbis) | AU (Sun/NeXT) |
|---|---|---|
| Container introduced | 2000 (Xiph.Org) | Early 1990s (Sun Microsystems) |
| Magic bytes | OggS (0x4f676753) |
.snd (0x2e736e64) |
| Endianness | Little-endian bitstream | Big-endian header and PCM |
| Default codec | Vorbis (lossy, MDCT) | 8-bit mu-law (legacy), 16-bit PCM (modern) |
| Typical bitrate | 96-320 kbps VBR | 64 kbps (8 kHz mu-law) up to 1411 kbps (44.1 kHz 16-bit stereo) |
| 1-minute file size | ~1 MB at 128 kbps | ~480 KB (8 kHz mu-law mono) to ~10 MB (44.1 kHz 16-bit stereo) |
| Metadata | Vorbis comments (UTF-8 tags) | Optional annotation chunk (ASCII, after header) |
| Browser playback | Chrome, Firefox, Edge, Opera (not Safari) | None natively |
| Native API support | HTML5 <audio>, Web Audio |
Java javax.sound.sampled.AudioFileFormat.Type.AU |
| Best for | Streaming, games, podcasts | Java apps, Unix system sounds, DSP test vectors |
| Sample Rate | Channels | Resulting bitrate (16-bit PCM) | Typical use |
|---|---|---|---|
| 8000 Hz | Mono | 128 kbps | Telephony, Solaris /dev/audio, voice prompts |
| 11025 Hz | Mono | 176 kbps | Old multimedia, low-bandwidth voice |
| 22050 Hz | Mono / Stereo | 353-705 kbps | Sound effects, Java game audio |
| 44100 Hz | Stereo | 1411 kbps | CD-quality music, default javax.sound playback |
| 48000 Hz | Stereo | 1536 kbps | Broadcast, DAW interchange, video sync |
Tip: since AU on this converter outputs uncompressed PCM, your output size scales linearly with sample rate and channel count — going from 44.1 kHz stereo to 8 kHz mono drops the file size by ~11x.
OGG Vorbis is lossy MDCT compression typically at 96-192 kbps; AU on this converter writes uncompressed linear PCM. A 3-minute Vorbis track at 128 kbps is about 2.9 MB; the same audio as 44.1 kHz 16-bit stereo PCM in an AU container is about 31.7 MB (≈1.41 Mbps × 180 s ÷ 8). To shrink the AU output, switch Audio Channel to Mono and Audio Sample Rate to 22050 Hz or 8000 Hz.
The output is signed linear PCM in big-endian AU containers (the Java-friendly modern variant). The AU specification supports many encodings — 8-bit mu-law (encoding 1), 8-bit A-law (encoding 27), 8/16/24/32-bit signed linear PCM (encodings 2-5), 32/64-bit IEEE float (encodings 6-7), plus various G.722/G.723 ADPCM modes (23-26) — but javax.sound.sampled and most modern tools default to 16-bit signed PCM, which is what you get here. If you specifically need 8-bit mu-law for SPARC-era /dev/audio compatibility, post-process with sox in.au -e mu-law -b 8 -r 8000 -c 1 out.au.
AudioSystem.getAudioInputStream() play these files?Yes. AudioFileFormat.Type.AU is one of five formats the JDK recognizes natively (the others are WAVE, AIFF, AIFC, SND), so you do not need a third-party SPI like JOrbis or Tritonus. Pick 44100 Hz Stereo for music assets and 22050 Hz Mono for short UI sounds and the output will load and play through any standard Java audio mixer.
The AU output target on xconvert is uncompressed linear PCM, so there is no encoder bitrate to tune — file size is fully determined by sample rate × bit depth × channel count × duration. The variables you can change are Audio Channel (Mono/Stereo), Audio Sample Rate (8000-48000 Hz), and Trim. For a compressed lossless alternative use OGG to FLAC; for a smaller lossy target use OGG to MP3.
No. Vorbis comments (UTF-8 key/value tags such as ARTIST, ALBUM, TITLE) do not have a direct equivalent in AU. AU only supports an optional ASCII annotation chunk between the header and the audio data, and most tools (Audacity, Java Sound, sox) ignore it. If you need tagged distribution files, convert to OGG to M4A or MP3 instead.
Yes — under Trim, switch from "Unchanged" to "Trim" and enter a Start Time and Duration. Values accept either plain seconds (12.5) or HH:MM:SS.sss (00:00:12.500). Because AU is uncompressed, trimming is the most effective way to keep the output small: a 10-second mono 22050 Hz clip is about 430 KB, while the full 4-minute track at the same settings is about 10 MB.
VLC and Audacity open AU on every platform. macOS QuickTime Player and Windows Media Player do not handle AU natively — on Windows you can register a codec or use ffplay from FFmpeg, on macOS use afplay (Core Audio handles AU) or sox. Browsers do not play AU natively; if you need web playback, target OGG to MP3 or OGG to WAV.
It is technically lossy because OGG Vorbis itself is lossy — the Vorbis decoder reconstructs an approximation of the original PCM, and that approximation is what gets written into the AU container. Going OGG → AU does not add a second generation of lossy encoding (linear PCM is lossless storage), but the output cannot have higher fidelity than the OGG input. For best results, start from the highest-bitrate OGG you have and pick 44100 Hz Stereo to avoid resampling artifacts.
Yes. Add as many .ogg files as you need in one batch and each is processed and downloadable independently. For format-mixing pipelines (mp3, flac, m4a, wma, oga sources alongside ogg), use the generic Audio to AU page instead, which accepts every supported audio extension in a single drop.