Convert AU to AIF — Free Online Tool
Convert Sun AU audio files to Apple AIF format using lossless PCM encoding — preserving every bit of the original audio data. Both formats use uncompressed PCM audio, so this conversion is a clean remux from a Unix legacy container to a Mac-native high-quality container with no quality loss.
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FFmpeg Command
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DownloadHow It Works
AU files store raw PCM audio in a simple header format developed by Sun Microsystems, while AIF is Apple's uncompressed audio container built around the same PCM data structure. During this conversion, FFmpeg reads the PCM audio stream from the AU container and rewraps it into the AIF container using the pcm_s16be codec — 16-bit big-endian signed PCM — which is natively supported by both formats. Since both AU and AIF can carry pcm_s16be audio, no re-encoding or sample conversion occurs; the raw audio samples are transferred directly. The result is a bit-perfect copy of the original audio, just wrapped in an AIF container that macOS, Logic Pro, GarageBand, and other Apple-ecosystem tools will recognize natively.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no audio data is sent to any server. |
-i input.au
|
Specifies the input AU file. FFmpeg reads the Sun AU container header to identify the audio codec (typically pcm_s16be, pcm_mulaw, or pcm_alaw) and stream parameters such as sample rate and channel count. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed codec for AIF files. Since AU files commonly carry pcm_s16be natively, this is often a direct stream copy in terms of sample format; for AU files with mu-law or A-law audio, FFmpeg decodes and re-encodes to this format. |
output.aif
|
Specifies the output filename with the .aif extension. FFmpeg uses the file extension to select the AIFF muxer, which wraps the PCM audio stream in Apple's Audio Interchange File Format container — making the file natively compatible with macOS, Logic Pro, GarageBand, and other Apple audio tools. |
Common Use Cases
- Importing legacy Unix workstation audio recordings or NeXT computer sound files into macOS audio editing software like Logic Pro or GarageBand, which prefer AIF over AU
- Migrating an archive of early internet audio samples (often distributed as .au files in the 1990s) to AIF for long-term preservation on Apple systems
- Preparing Sun AU audio assets from a Unix-based game or application for use in an Apple-platform port, where AIF is the standard uncompressed audio format
- Converting telephony audio samples stored as AU with pcm_alaw or pcm_mulaw encoding — first normalizing to pcm_s16be in AIF for use in modern audio production workflows
- Transferring audio files from a Unix/Linux development environment to a Mac post-production environment without any quality loss or resampling artifacts
Frequently Asked Questions
No — this conversion is completely lossless. Both AU and AIF support 16-bit big-endian PCM (pcm_s16be), so the raw audio samples are transferred directly from one container to the other without any re-encoding. The output AIF file is a bit-perfect copy of the audio data in the original AU file.
In most cases the file sizes should be nearly identical, since both formats store the same uncompressed PCM samples. AIF does add slightly more metadata in its header structure (using Apple's AIFF chunk format), but the difference is typically only a few hundred bytes. If the AU file used a compressed codec like pcm_mulaw or pcm_alaw, the AIF output will be larger because FFmpeg decodes it to full 16-bit pcm_s16be during conversion.
Yes. AU files frequently carry telephony-grade audio encoded with pcm_mulaw (mu-law) or pcm_alaw (A-law), which compress audio to 8-bit. FFmpeg will decode that audio and re-encode it as 16-bit pcm_s16be in the AIF output. The resulting audio will be louder and cleaner-looking in a waveform view, but the original quality ceiling of the mu-law or A-law source still applies — converting to AIF does not recover any fidelity lost at the time of the original encoding.
Yes — AIF natively supports pcm_s24be (24-bit) and pcm_s32be (32-bit) as well as 32-bit and 64-bit floating-point PCM. However, the AU format's most common codec is pcm_s16be, so the default conversion targets 16-bit to match the source. If your AU file was recorded at a higher effective quality, you can modify the FFmpeg command to use '-c:a pcm_s24be' for a 24-bit AIF output, though you won't gain real additional fidelity from a 16-bit source.
AU files have a very minimal header with almost no metadata support — the format predates rich tagging standards and typically stores only a text annotation field. AIF supports ID3-style tags via its ID3 chunk, but because the AU source carries essentially no standardized metadata, the AIF output will also have minimal tag information. You would need to add tags manually after conversion using a tool like iTunes, Mp3tag, or Kid3.
Replace 'pcm_s16be' in the command with another AIF-compatible codec. For example, use '-c:a pcm_s24be' for 24-bit audio or '-c:a pcm_f32be' for 32-bit floating-point audio. The full modified command would look like: ffmpeg -i input.au -c:a pcm_s24be output.aif. Note that AIF does not support lossy codecs like AAC or MP3 — all valid options are uncompressed PCM variants.
Technical Notes
Both AU and AIF are uncompressed PCM container formats that share the pcm_s16be codec as their common default, making this one of the cleanest possible audio container conversions. The AU format uses a minimal fixed-size or variable-size header followed by raw audio data, with no support for chapters, multiple tracks, or embedded artwork. AIF uses Apple's IFF-derived chunk structure, which is more extensible — it can store sample loop points (useful for audio instruments), marker data, and ID3 tags — but none of this metadata originates in an AU source file. One important edge case: AU files encoded with 8-bit pcm_mulaw or pcm_alaw (common in telephony and early Unix systems) will be decoded to 16-bit linear PCM in the AIF output, which doubles the file size relative to the source while preserving the original audio fidelity. The '-b:a' bitrate parameter has no meaningful effect on uncompressed PCM codecs; quality is entirely determined by sample rate and bit depth, both of which are preserved from the source by default.