Extract Audio from MP4 to AU — Free Online Tool
Extract audio from MP4 video files and save it as AU format — Sun's legacy Unix audio container — using lossless PCM 16-bit big-endian encoding. This tool is ideal for Unix/Linux audio pipelines and legacy system compatibility where the simple, headerless-adjacent AU structure is required.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
During this conversion, FFmpeg discards the MP4's video stream entirely and re-encodes the audio — whether it was originally AAC, MP3, or Opus inside the MP4 — into PCM signed 16-bit big-endian audio (pcm_s16be), which is the AU format's default and most widely supported codec. This is a full audio decode-and-re-encode operation: the compressed audio from the MP4 is first fully decoded to raw PCM samples, then written into the AU container with its minimal 28-byte header. Because the output is uncompressed PCM, there is no generational quality loss relative to the original source signal, but file sizes will be substantially larger than the compressed AAC or MP3 audio in the source MP4.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion both in the browser via WebAssembly and locally on your desktop. |
-i input.mp4
|
Specifies the input file — an MP4 container that may contain video, compressed audio (commonly AAC), subtitles, and chapter data, all of which FFmpeg will parse before the output flags determine what to keep. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the video stream from the MP4 and produce an audio-only output — essential since the AU format has no capacity to store video data. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, decoding whatever compressed audio codec was in the MP4 (typically AAC) into uncompressed samples stored in the byte order native to the AU format's Sun SPARC origins. |
output.au
|
Defines the output filename with the .au extension, which FFmpeg uses to confirm the Sun AU container format and write the 28-byte AU header followed by the raw pcm_s16be sample data. |
Common Use Cases
- Feeding extracted audio into Unix or Linux audio processing pipelines that expect the native AU format, such as legacy Sun workstation tools or SoX-based scripts
- Preparing audio content for playback in older Java applications, which historically had built-in AU format support via the javax.sound API
- Archiving the audio track of an MP4 recording in an uncompressed, structurally simple format where long-term readability without proprietary decoders is a priority
- Supplying raw PCM audio in AU containers to scientific or signal-processing software that reads AU files as a lightweight wrapper around uncompressed sample data
- Converting MP4 audio to AU for use in early internet or retro web projects where .au files were the standard browser-embeddable audio format in the 1990s
- Extracting dialogue or sound effects from MP4 video assets into AU files for import into Unix-based digital audio workstations or sampling tools that prefer uncompressed input
Frequently Asked Questions
The conversion involves decoding the compressed audio codec inside your MP4 — typically AAC — to raw PCM samples, then writing those samples as pcm_s16be in the AU file. This decode step introduces the same minimal loss as any AAC decode, but no additional compression is applied; the AU output is uncompressed. If your source MP4 used high-quality AAC at 192k or above, the resulting AU file will be a faithful uncompressed representation of that decoded audio.
The MP4's audio track is stored as compressed AAC, MP3, or Opus, which can be 10–20x smaller than uncompressed audio. The AU format stores raw PCM samples with no compression — stereo audio at 44.1kHz in 16-bit PCM produces roughly 10MB per minute. Stripping the video stream reduces size compared to the full MP4, but the audio portion alone will be much larger than the compressed audio stream inside the MP4.
PCM s16be stands for Pulse-Code Modulation, signed 16-bit, big-endian byte order. Big-endian byte ordering was the native format of Sun SPARC workstations, where the AU format originated, so pcm_s16be became the canonical AU codec. It provides CD-quality bit depth and is universally readable by any software that supports AU, making it the most compatible choice for this format.
No. The AU format has an extremely minimal header — just 28 bytes covering magic number, data offset, data size, encoding type, sample rate, and channel count. It has no standardized fields for ID3-style metadata, chapters, or multiple audio tracks. All metadata present in the MP4 container, including titles, artist tags, and chapter markers, will be lost in the conversion.
You can add '-ar 22050' before the output filename to resample to 22050Hz, for example, which was common for early AU files on the web. To use 8-bit encoding instead of 16-bit, replace '-c:a pcm_s16be' with '-c:a pcm_u8' or '-c:a pcm_mulaw' for mu-law compressed audio, which is another codec the AU container natively supports. The full modified command would look like: ffmpeg -i input.mp4 -vn -ar 22050 -c:a pcm_mulaw output.au
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.mp4; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mp4}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.mp4) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au"'. This applies the same extraction and pcm_s16be encoding to every MP4 in the current directory, which is especially useful when the browser tool's 1GB limit is a constraint for large files.
Technical Notes
The AU format (magic number 0x2E736E64, '.snd') uses a fixed 28-byte header followed by raw audio data, making it one of the simplest audio container formats in existence. Because it lacks a compression layer, the pcm_s16be codec used here stores two bytes per sample per channel — a stereo 44.1kHz file consumes approximately 176,400 bytes per second. The AU format does not support multiple audio tracks, so if your MP4 contains multiple audio streams, FFmpeg will extract only the first one by default; you can target a specific stream with '-map 0:a:1' for the second audio track, for example. Subtitle tracks, chapter markers, and video streams from the MP4 are all silently dropped. The big-endian byte order of pcm_s16be means AU files are not byte-for-byte identical to WAV PCM files (which use little-endian pcm_s16le), a distinction that matters for low-level binary processing. If playback compatibility on modern systems is the goal, WAV or FLAC would be more practical choices; AU is best chosen specifically for Unix legacy workflows or Java sound API compatibility.