Extract Audio from M4V to AU — Free Online Tool
Extract audio from M4V files and save it as AU format — a raw PCM audio format developed by Sun Microsystems. This tool strips the AAC or MP3 audio track from your M4V container and re-encodes it as uncompressed 16-bit big-endian PCM, producing a lossless AU file suitable for Unix audio pipelines and legacy systems.
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FFmpeg Command
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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
M4V files from iTunes or iOS typically carry AAC-encoded audio wrapped in an MPEG-4 container alongside an H.264 or H.265 video stream. This conversion discards the video stream entirely and decodes the compressed AAC audio, then re-encodes it as uncompressed PCM (specifically pcm_s16be — signed 16-bit samples in big-endian byte order) and writes it into a Sun AU container. Because AAC is a lossy format and AU/PCM is uncompressed, the output will be a faithful but slightly larger representation of the already-lossy AAC audio — there is no quality gain from the decompression step, but no further lossy compression is applied.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — your M4V file never leaves your device. |
-i input.m4v
|
Specifies the input file — your M4V source, which is an MPEG-4 container typically holding H.264 or H.265 video alongside AAC-compressed audio, as used by iTunes and iOS devices. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the H.264 or H.265 video stream in the M4V. Since AU is a pure audio format with no video container support, this flag is required to produce a valid output file. |
-c:a pcm_s16be
|
Decodes the AAC audio from the M4V and re-encodes it as signed 16-bit big-endian PCM — the default and most faithful audio codec for the Sun AU format, matching the byte-order convention of the original Unix/SPARC systems where AU was developed. |
output.au
|
Specifies the output filename with the .au extension, which tells FFmpeg to wrap the PCM audio data in a Sun AU container with the appropriate .snd magic number header. |
Common Use Cases
- Feeding audio from an iTunes M4V download into a Unix or Linux audio processing pipeline that expects raw AU input, such as legacy SoX-based scripts.
- Archiving the audio track of an iOS-purchased M4V in an uncompressed format for long-term preservation without introducing additional lossy encoding stages.
- Supplying uncompressed PCM audio from an M4V to audio analysis or scientific tools (e.g., signal processing or speech research software) that read AU files natively.
- Stripping and converting M4V audio for use in early Java applications or applets, which historically used AU as their native audio format via the AudioClip API.
- Preparing audio from M4V tutorials or lectures for ingestion into a legacy Unix workstation environment where AU is the standard audio interchange format.
- Debugging or inspecting the raw audio content of an M4V file without any compression artifacts introduced by a secondary encoding step.
Frequently Asked Questions
The conversion does not introduce any new lossy compression — AU with pcm_s16be is uncompressed PCM audio. However, the source AAC audio inside the M4V was already encoded with lossy compression when the M4V was created, so that original quality loss cannot be recovered. What you get in the AU file is the best possible representation of the AAC audio decoded into raw form, with no further degradation from this conversion step.
AU stores audio as raw uncompressed PCM samples (16 bits per sample, per channel), while the AAC audio inside your M4V is heavily compressed — typically achieving 10:1 or greater compression ratios. When AAC is decoded to PCM for the AU container, the audio data expands significantly. For example, a 128 kbps AAC stereo track becomes roughly 1,411 kbps of uncompressed PCM, so a 100 MB M4V with 10 minutes of audio could produce an AU file of 100 MB or more for audio alone.
No. The Sun AU format has an extremely simple header structure and supports only a single audio stream with no metadata containers for chapters, track names, or multiple audio tracks. Any chapter markers, alternate language audio tracks, or iTunes metadata present in your M4V will be lost in the conversion. If you need to preserve chapters or select a specific audio track, you should pre-select the desired track before conversion or use a richer output format.
Yes — the AU format also supports pcm_s8 (8-bit signed), pcm_u8 (8-bit unsigned), pcm_alaw (G.711 A-law, common in telephony), and pcm_mulaw (G.711 mu-law). You can substitute any of these in the FFmpeg command by replacing 'pcm_s16be' with your preferred codec, for example: ffmpeg -i input.m4v -vn -c:a pcm_mulaw output.au. The mu-law and A-law codecs produce smaller files and are lossy but designed for voice audio, while pcm_s16be gives you the highest fidelity.
M4V files — especially iTunes purchases — sometimes include multiple audio tracks for different languages or commentary. To select a specific track, add the '-map' flag to the FFmpeg command before the output filename. For example, to extract the second audio track: ffmpeg -i input.m4v -vn -map 0:a:1 -c:a pcm_s16be output.au. The index 0:a:1 means the first input file, audio stream index 1 (zero-based). Run ffmpeg -i input.m4v first to see a list of all audio streams and their indices.
On Linux or macOS, you can loop over all M4V files in a directory with: for f in *.m4v; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.m4v}.au"; done. On Windows Command Prompt, use: for %f in (*.m4v) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au". Note that AU files are uncompressed, so batch-converting a large M4V library will consume significant disk space — plan accordingly.
Technical Notes
The Sun AU format uses a minimal fixed header (24 bytes minimum) containing a magic number (.snd), data offset, data size, encoding type, sample rate, and channel count. FFmpeg writes the pcm_s16be encoding descriptor (value 3 in the AU specification) for the default output. Big-endian byte ordering is intrinsic to the format's Unix workstation heritage (SPARC architecture). Because AU has no concept of ID3 tags, iTunes metadata (artist, album, track title, artwork) embedded in the M4V's MPEG-4 container is completely discarded. Sample rate and channel count from the source AAC stream are preserved in the AU header, so if your M4V contains 44.1 kHz stereo AAC, the output AU file will also be 44.1 kHz stereo. One subtle limitation: the AU data size field is a 32-bit value, which means files exceeding approximately 4 GB (around 6.5 hours of 44.1 kHz stereo PCM) may have header size field overflow issues in some legacy readers, though FFmpeg itself handles this gracefully by writing 0xFFFFFFFF as the size sentinel.