Convert OGA to AU — Free Online Tool
Convert OGA (Ogg Audio) files to Sun AU format, decoding Vorbis or FLAC streams and re-encoding them as raw PCM audio in big-endian 16-bit format. This conversion bridges modern open-source audio containers with a legacy Unix format still used in Java applications, certain Unix/Linux toolchains, and early internet audio workflows.
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FFmpeg Command
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DownloadHow It Works
OGA files typically contain Vorbis (lossy) or FLAC (lossless) compressed audio streams inside an Ogg container. The Sun AU format, by contrast, only supports uncompressed PCM audio — it has no provision for compressed codecs. During this conversion, FFmpeg fully decodes the OGA audio stream (whether Vorbis or FLAC) to raw PCM samples, then re-encodes them as 16-bit big-endian PCM (pcm_s16be) and writes them into the minimal AU container with its fixed 24-byte header. This means the output is always uncompressed, regardless of whether your source OGA used lossy Vorbis or lossless FLAC. If your source was Vorbis, any compression artifacts introduced during the original Vorbis encoding are baked into the output — the AU file will be uncompressed but not 'restored' to lossless quality.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. This conversion runs the same FFmpeg engine locally in your browser via WebAssembly (FFmpeg.wasm), so no files leave your machine. |
-i input.oga
|
Specifies the input file — an OGA container that may hold Vorbis, FLAC, or Opus audio. FFmpeg automatically detects the codec inside the Ogg container and selects the appropriate decoder. |
-c:a pcm_s16be
|
Sets the output audio codec to signed 16-bit big-endian PCM, which is the standard and most widely compatible encoding for Sun AU files. This fully decodes the compressed OGA audio stream and writes it as raw, uncompressed PCM samples in the byte order expected by the AU format and Java's audio APIs. |
output.au
|
Specifies the output filename with the .au extension. FFmpeg uses this extension to automatically select the Sun AU container format (also known as the SND format), wrapping the pcm_s16be audio data in the minimal AU file header. |
Common Use Cases
- Loading audio into Java applications or applets that rely on the javax.sound.sampled API, which has native support for the AU format but not for Ogg Vorbis
- Using OGA audio files in legacy Unix/Linux audio pipelines or tools (such as older versions of SoX or Sun's audio utilities) that expect AU-formatted input
- Preparing audio for playback on classic workstations or emulated Sun/NeXT environments where AU is the native audio format
- Converting open-format OGA audio to a simple, headerless-compatible PCM container for embedding in scientific or engineering software that reads AU files as a standard input format
- Archiving or exporting audio from Ogg-based projects into an uncompressed format for waveform analysis, where the AU container's simplicity makes byte-level inspection straightforward
- Providing audio assets in AU format for web servers or early streaming setups that used AU files for short audio clips before MP3 and Ogg became dominant
Frequently Asked Questions
No — converting a Vorbis-encoded OGA file to AU does not recover quality lost during Vorbis compression. The Vorbis codec is lossy, meaning audio information was permanently discarded when the OGA file was originally created. Decompressing it to PCM for the AU file simply unpacks the already-degraded signal into uncompressed form. If your OGA source used FLAC (a lossless codec), the AU output will be a true lossless representation of the original audio.
OGA files use compressed audio codecs — Vorbis can achieve compression ratios of 5:1 to 10:1 or more, and FLAC typically compresses to around 50–60% of raw PCM size. The AU format stores audio as raw, uncompressed 16-bit PCM samples at the full sample rate, so a 4-minute OGA file that might be 4MB as Vorbis could become 40MB or more as AU. This is expected and inherent to the conversion, not a problem with the tool.
No. The Sun AU format has an extremely minimal header — just 24 bytes containing sample rate, channel count, encoding type, and an optional free-form annotation field. It has no standardized support for ID3 tags or Vorbis comment-style metadata. Any artist, album, title, or track metadata embedded in your OGA file will be lost during this conversion. If preserving metadata is important, consider an intermediate format like FLAC or WAV before archiving.
pcm_s16be stands for PCM, signed, 16-bit, big-endian — the default and most compatible audio encoding for AU files. 'Big-endian' refers to the byte order used by the original Sun SPARC architecture that defined the format. AU also supports pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw. You can substitute a different codec by changing the flag — for example, use '-c:a pcm_mulaw' for telephony-style G.711 mu-law encoding, which produces smaller files but with lower dynamic range.
On Linux or macOS, you can use a shell loop: 'for f in *.oga; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.oga}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.oga) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. This runs the same conversion command against every OGA file in the current directory, outputting a correspondingly named AU file for each.
Yes. By default, FFmpeg preserves the sample rate and channel count from the source OGA file. You can override these by adding '-ar 44100' to set a specific sample rate (e.g., 44.1 kHz) or '-ac 1' to convert stereo to mono. For example: 'ffmpeg -i input.oga -c:a pcm_s16be -ar 8000 -ac 1 output.au' would produce a mono, 8 kHz AU file — a common configuration for telephony or legacy system compatibility.
Technical Notes
The Sun AU format is one of the oldest digital audio container formats still encountered in active use, primarily because Java's standard library (javax.sound.sampled) includes built-in AU decoding support — making it a practical delivery format for Java-based applications and applets. The format's structure is intentionally minimal: a magic number (0x2e736e64, or '.snd'), a fixed header describing data offset, data size, encoding, sample rate, and channel count, followed by raw audio data. There are no provisions for chapters, multiple audio tracks, or metadata — all of which OGA supports. The pcm_s16be encoding used by default provides a 96 dB dynamic range and supports any standard sample rate, but its big-endian byte order means the files are not directly interchangeable with little-endian formats like WAV's PCM without byte-swapping. If your OGA source used the libopus codec (Opus audio), the same transcoding pipeline applies — Opus is decoded to PCM first, then written as pcm_s16be. Note that Opus internally operates at 48 kHz, so if your source sample rate differs, FFmpeg will resample accordingly during conversion. AU files produced this way are fully streamable, as the format was designed for network audio delivery on early Unix systems.