Convert AVI to AU — Free Online Tool
Convert AVI video files to AU audio format, extracting the audio stream and re-encoding it as 16-bit big-endian PCM — the native encoding of Sun's AU format used on Unix systems and early internet applications. This tool strips the video entirely and outputs a raw, uncompressed audio file compatible with legacy Unix software and Java's AudioInputStream.
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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
AVI files typically carry compressed audio streams encoded in formats like MP3 (libmp3lame) or AAC. Since the AU format only supports raw PCM audio codecs and has no concept of video tracks, this conversion performs two distinct operations: the video stream is completely discarded, and the audio stream is fully decoded and re-encoded as pcm_s16be — 16-bit signed PCM with big-endian byte order. This is a lossy-to-lossless step in terms of PCM representation, but the original quality ceiling is limited by whatever lossy codec was used in the source AVI. The output AU file contains a minimal fixed-length header followed by raw audio samples, with no support for metadata, chapters, or multiple audio tracks.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. This command runs entirely in your browser via FFmpeg.wasm (WebAssembly) — no files leave your device. |
-i input.avi
|
Specifies the input AVI file. FFmpeg reads the AVI container and demuxes its audio and video streams; the video stream will be automatically discarded since the AU output format cannot carry video data. |
-c:a pcm_s16be
|
Sets the audio codec to pcm_s16be — signed 16-bit big-endian PCM. This is the default and most compatible encoding for the AU format, reflecting its origin on big-endian Sun SPARC hardware, and it decodes whatever compressed audio (MP3, AAC, etc.) was in the AVI into raw, uncompressed PCM samples. |
output.au
|
Specifies the output file with the .au extension, which tells FFmpeg to use the Sun AU muxer. The resulting file will have AU's minimal fixed-length header followed by the raw 16-bit big-endian PCM audio data, ready for playback on Unix systems or use in Java audio applications. |
Common Use Cases
- Extracting dialogue or soundtrack audio from an AVI video to feed into a Java application that uses javax.sound.sampled, which natively reads AU files via AudioInputStream
- Preparing audio from legacy AVI recordings for playback on Unix workstations or older Sun Solaris systems where AU is the system-native sound format
- Converting AVI audio to a raw PCM AU file for use as input in audio processing pipelines or DSP tools that expect headerless or minimally-headered PCM streams
- Archiving the audio component of an AVI file in an uncompressed format for long-term preservation, where AU's simple structure ensures readability without proprietary decoders
- Generating AU audio samples from AVI source footage for embedding in early-web or retro HTML pages that used the AU format for browser audio via the bgsound or embed tags
- Supplying audio content from an AVI clip to telephony or VoIP testing tools that accept pcm_s16be or AU-wrapped PCM as their audio input format
Frequently Asked Questions
It depends on the audio codec in your source AVI. If the AVI contains lossy audio — which is typical, since AVI commonly uses MP3 or AAC — that quality loss already exists and cannot be recovered. The conversion re-encodes the decoded audio as pcm_s16be, which is lossless PCM, so no additional quality is lost beyond what the original AVI encoding already introduced. If your AVI somehow carried uncompressed PCM audio, the output AU will be a faithful representation of the original samples.
The AVI's audio was almost certainly stored in a compressed format like MP3 or AAC, which achieves significant file size reduction. The AU output stores audio as uncompressed 16-bit PCM samples, which requires roughly 1.4 MB per minute per channel at 44.1 kHz. A 128 kbps MP3 track expanded to PCM will be roughly 8–10 times larger in the AU file, which is expected behavior for any compressed-to-uncompressed audio conversion.
No. The AU format uses a fixed minimal header that contains only technical fields: a magic number, data offset, data size, encoding type, sample rate, and channel count. There is no provision for ID3 tags, XMP metadata, or any textual annotations. Any title, artist, or comment metadata present in the source AVI will be completely dropped during this conversion.
Yes. The AU format supports several PCM encodings including pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw. To use one, replace '-c:a pcm_s16be' in the FFmpeg command with your preferred codec, such as 'ffmpeg -i input.avi -c:a pcm_mulaw output.au'. The mu-law (G.711) encoding is particularly useful for telephony applications, while 8-bit PCM significantly reduces file size at the cost of dynamic range.
The video stream is completely ignored and not written to the output. AU is a pure audio-only format with no container support for video, so FFmpeg automatically drops the video stream when writing to the AU muxer. You do not need to pass '-vn' explicitly — the output format itself enforces audio-only output. No video frames are decoded, which makes the conversion relatively fast regardless of video resolution.
On Linux or macOS, you can use a shell loop: 'for f in *.avi; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.avi}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.avi) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. This applies the same pcm_s16be encoding to every AVI in the current directory, outputting one AU file per input. The browser-based tool on this page processes one file at a time, so the FFmpeg command is the recommended approach for bulk conversions or files larger than 1 GB.
Technical Notes
The AU format (.au, also sometimes called SND) was developed by Sun Microsystems and became the default audio format on Unix platforms in the late 1980s and 1990s. Its header is intentionally minimal: an 8-byte magic number (.snd), a 4-byte data offset, data size, encoding type, sample rate, and channel count — nothing more. The pcm_s16be encoding specified in this conversion stores samples as signed 16-bit integers in big-endian byte order, which reflects AU's origin on big-endian SPARC hardware. Notably, AU's big-endian byte order differs from WAV's little-endian PCM, so the two are not byte-for-byte interchangeable even when both store 16-bit PCM. The AU format supports a maximum of one audio track, so if your AVI contains multiple audio streams, only the first (default) audio stream will be extracted. Sample rates and channel counts from the source audio are preserved in the AU header as-is. There is no resampling or channel downmixing applied unless explicitly added to the command. Java's standard library has built-in AU support through javax.sound.sampled, making this format particularly relevant for Java-based audio applications.