Extract Audio from AVI to AU — Free Online Tool
Extract audio from AVI video files and save it as AU format — a simple PCM-based audio container developed by Sun Microsystems for Unix systems. The conversion drops the video stream entirely and re-encodes the audio to 16-bit big-endian PCM (pcm_s16be), producing an uncompressed, lossless audio file.
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FFmpeg Command
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DownloadHow It Works
AVI files typically store audio in compressed formats such as MP3 (libmp3lame), AAC, or Vorbis. When converting to AU, FFmpeg discards the video stream completely using the -vn flag, then decodes the compressed audio and re-encodes it to pcm_s16be — 16-bit signed big-endian PCM — which is the standard codec for the AU container. This means the audio is fully decompressed from its lossy source and written as raw PCM samples. Because the source audio in AVI is almost certainly lossy (MP3 or AAC), the output is technically lossless PCM, but it cannot recover quality lost in the original compression — what you get is a lossless representation of the already-decoded audio. The resulting AU file has a minimal fixed-size header followed by raw audio data, making it straightforward to parse but larger than the compressed original.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which here runs entirely in your browser via WebAssembly (FFmpeg.wasm) — no data leaves your machine. |
-i input.avi
|
Specifies the input AVI file. FFmpeg reads the container and identifies the interleaved audio and video streams inside, which may include MP3, AAC, or Vorbis audio alongside an H.264 or MJPEG video track. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the video stream from the AVI. Since AU is a pure audio format with no video support, this flag is essential — without it, FFmpeg would attempt to handle the video stream and likely error out. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, which is the standard and default codec for AU files. This decodes whatever compressed audio (MP3, AAC, etc.) was in the AVI and re-encodes it as uncompressed PCM samples in the byte order required by the Sun AU format. |
output.au
|
Specifies the output filename with the .au extension, which tells FFmpeg to use the Sun AU container format. The combination of this container and the pcm_s16be codec produces a valid, broadly compatible AU file readable by Java applications, Unix audio tools, and legacy Sun/NeXT software. |
Common Use Cases
- Feeding audio from legacy AVI recordings into Unix or Linux audio processing pipelines that expect AU format as input
- Extracting dialogue or narration from old AVI training videos to load into Sun Microsystems-era or NeXT-based audio tools that only read AU files
- Preparing audio assets from AVI source footage for use in Java applications, which have native AU/pcm_s16be support through the javax.sound API
- Archiving the audio track from AVI home videos as uncompressed PCM in AU format for long-term preservation without codec dependency
- Stripping audio from AVI files to create raw PCM AU samples for use in academic signal processing or audio analysis software on Unix workstations
- Converting AVI audio tracks to AU format for compatibility with early internet streaming setups or web servers configured to serve .au audio files
Frequently Asked Questions
No — the output will be lossless PCM, but it cannot restore quality that was already lost. AVI files commonly store audio as MP3 or AAC, both of which are lossy codecs. FFmpeg decodes that compressed audio and writes it out as uncompressed pcm_s16be in the AU file, which is a perfect lossless copy of the decoded signal. The AU file will be larger than the original compressed audio, but it won't sound better than what the AVI's lossy codec already produced.
AVI files typically store audio in compressed formats like MP3 (often at 128k–320k bitrate), which use perceptual encoding to reduce file size significantly. The AU format with pcm_s16be stores every audio sample as raw uncompressed data — for stereo audio at 44.1kHz, that's roughly 10MB per minute. A 5-minute AVI with 128k MP3 audio might have under 5MB of audio, while the equivalent AU file could be 50MB or more. This is expected behavior, not an error.
Yes, the AU format supports multi-channel audio including stereo, and pcm_s16be will preserve the channel layout from your AVI's audio track. However, AU has no support for multiple audio tracks — if your AVI file contains more than one audio track, only the first (default) audio track will be extracted into the AU file. If you need a specific track, you would need to modify the FFmpeg command to add a stream selector like -map 0:a:1 to pick the second audio track.
Yes, the AU format supports several PCM codecs beyond the default pcm_s16be, including pcm_s8 (8-bit signed), pcm_u8 (8-bit unsigned), pcm_alaw (G.711 A-law), and pcm_mulaw (G.711 mu-law). You can substitute any of these by replacing pcm_s16be in the command, for example: ffmpeg -i input.avi -vn -c:a pcm_mulaw output.au. The pcm_alaw and pcm_mulaw options produce smaller files and were commonly used in telephony applications. The default pcm_s16be offers the best audio fidelity.
You can batch process multiple AVI files using a shell loop. On Linux or macOS, run: for f in *.avi; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.avi}.au"; done. On Windows Command Prompt: for %f in (*.avi) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au". This iterates over every AVI in the current directory and produces a matching AU file for each one.
No — the AU format has an extremely minimal header that includes only sample rate, channel count, encoding type, and an optional free-text annotation field. It does not support structured metadata tags like title, artist, or album. Any ID3 or INFO tags stored in your AVI file will be discarded during this conversion. If metadata preservation matters, AU is not an appropriate target format; consider formats like FLAC or WAV instead.
Technical Notes
The AU format originated with Sun Microsystems and NeXT, and remains notable for its big-endian byte order — unlike WAV which is little-endian. The pcm_s16be codec writes 16-bit signed samples in big-endian order, which is natively correct for AU and means no byte-swapping is needed on SPARC or PowerPC architectures. AVI files with variable bitrate (VBR) MP3 audio may occasionally cause minor timing discrepancies during decode, though these are generally inaudible in the extracted PCM. Because the AU format does not support chapters, subtitle tracks, or multiple audio streams, all such data from the source AVI is silently dropped — the -vn flag handles video, and any secondary audio tracks are ignored by default. The resulting AU file will match the sample rate and channel count of the source audio without resampling unless you explicitly add -ar or -ac flags. One important limitation: AU files have no widely standardized loudness normalization or gain metadata, so if your AVI audio is quiet, the AU output will be equally quiet — you would need to add a loudnorm or volume filter to the FFmpeg command to adjust levels.