AIFF to AU Converter

Convert AIFF to AU: What This Tutorial Covers

This guide is for anyone who needs to feed audio into a legacy Unix, Solaris, or Java pipeline that specifically reads Sun/NeXT .au files. AIFF and AU are both old, big-endian, uncompressed-PCM families, so at the default 16-bit PCM setting this conversion is essentially a lossless re-wrap — the same audio samples, repackaged from Apple's IFF-style header into Sun's 24-byte header — not a quality or size change.

How to Convert AIFF to AU

1. Upload Your AIFF File: Drag and drop your .aiff or .aif file onto the page, or click "+ Add Files" to browse. Several files can be queued and converted in one batch with shared settings.
2. Set Audio Sample Rate: Leave Audio Sample Rate on "Original" to keep the source rate (a true lossless re-wrap), or pick a target such as 44100 Hz, 48000 Hz, 16000 Hz, or 8000 Hz if a legacy tool expects a specific rate.
3. Set Audio Channel (Optional): Leave Audio Channel on "Original" to preserve mono/stereo, or downmix stereo to mono for a smaller voice file. Use the Trim control to export only a portion of the track.
4. Convert and Download: Click "Convert" and download your AU file. No sign-up, no watermark.

Walk-through: Getting the Right AU for Your Pipeline

The output AU is written as PCM_S16BE — 16-bit big-endian linear PCM — by default. That is the closest direct match to a typical 16-bit AIFF, which is also big-endian PCM, so leaving everything on "Original" gives you a byte-for-byte equivalent waveform in a Sun-style container. Choose your settings around the tool you are feeding:

• Modern Java (javax.sound.sampled / AudioSystem): keep "Original" sample rate and channel. 16-bit PCM AU loads cleanly.
• Classic Java applet sound (java.applet.AudioClip): that interface was built around 8-bit, 8000 Hz, mono μ-law clips. Set Audio Sample Rate to 8000 Hz and Audio Channel to Mono for the most compatible result.
• Solaris / SunOS /dev/audio or SoX scripts: keep CD-quality 44100 Hz unless the receiving stage downsamples; SoX reads 16-bit PCM AU natively.
• Scientific / signal-processing input (Octave, Csound, ASR corpora): match whatever sample rate the model or script documents — often 16000 Hz mono for speech.

Common Errors and How to Fix Them

• "The AU file won't play on my phone or in Windows Media Player" — AU is a legacy format with little native support outside Unix; this is expected. Play it in VLC, foobar2000, or Audacity, or convert it to a modern format instead.
• "My old Java applet plays only silence or noise" — the applet likely expects 8-bit μ-law at 8000 Hz mono; re-run the conversion with Audio Sample Rate set to 8000 Hz and Audio Channel set to Mono.
• "The AU file is the same size as the AIFF" — that is correct at the default 16-bit PCM. Both are uncompressed, so the payload is identical; only the small header differs. Pick a lower sample rate, downmix to mono, or use a compressed format to shrink it.
• "Samples sound garbled / byte-swapped" — this usually means the receiving software assumed little-endian. AU and 16-bit AIFF are both big-endian; the AU header (PCM_S16BE) declares this correctly, so use a player that honors the header.

When This Doesn't Work

If you actually need a smaller file rather than a legacy container, AU at default PCM will not help — it is uncompressed. Convert to AIFF to MP3 for a small, universally playable lossy copy, or AIFF to FLAC for lossless compression at roughly half the size. If you want a modern uncompressed PCM container that everything reads, use AIFF to WAV instead. AU only makes sense when a specific old Unix, Solaris, or Java tool requires the .au/snd format on its input.

Frequently Asked Questions

Is converting AIFF to AU lossless?

Yes, at the default settings. Both AIFF and AU are uncompressed big-endian PCM families, so writing a 16-bit AIFF to AU at PCM_S16BE with "Original" sample rate and channel keeps the audio samples byte-for-byte equivalent in value — only the container header changes. Loss only occurs if you downsample, downmix, or pick a μ-law/A-law output.

Will the AU file be smaller than the AIFF?

No, not at the default. AU written as 16-bit PCM is uncompressed just like AIFF, so the payload is the same size; the two headers differ by only a few dozen bytes. AU's classic 8-bit μ-law mode at 8000 Hz mono is roughly an 8x reduction over CD-quality, but that is lossy. For a genuinely smaller file without that telephony quality, use AIFF to FLAC or AIFF to MP3.

Why would I convert AIFF to AU at all in 2026?

The realistic reason is compatibility with software that predates modern formats: legacy Java applet sound, older Solaris/SunOS toolchains, and scientific or signal-processing scripts that read Sun's .au natively. If you do not have one of those constraints, WAV, FLAC, or MP3 are better general-purpose choices.

What codec does the AU output use, and is it big-endian?

The converter writes AU as PCM_S16BE — 16-bit linear PCM in big-endian byte order — by default. AU stores all fields, including sample data, in big-endian, which matches AIFF's byte order, so a 16-bit AIFF maps directly into AU without re-ordering the samples. In our testing, a 16-bit/44.1 kHz stereo AIFF and its AU output had identical PCM payloads, differing only in the header.

What sample rate should I use for a legacy Java AudioClip?

The classic java.applet.AudioClip interface was designed around 8-bit, 8000 Hz, mono μ-law AU files. If you are feeding old applet code or a textbook example, set Audio Sample Rate to 8000 Hz and Audio Channel to Mono. Newer javax.sound.sampled code reads 16-bit PCM AU as well, so for modern Java you can keep "Original."

How are my uploaded files handled?

Files are uploaded over an encrypted connection, processed on our servers, and deleted automatically a few hours after conversion — no sign-up, no watermark, never shared or made public. To go the other way, use AU to AIFF.