MyMediaTools

Convert AIFC to AU — Free Online Tool

Convert AIFC audio files to Sun AU format using PCM 16-bit big-endian encoding — a natural fit since both formats natively use big-endian byte order. This tool runs entirely in your browser via FFmpeg.wasm, with no file uploads required.

FFmpeg Command

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Estimated output:

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How It Works

AIFC files store audio using a variety of codecs including uncompressed PCM variants (pcm_s16be, pcm_s24be, pcm_s32be) and compressed formats like A-law and μ-law. During conversion to AU, FFmpeg decodes the AIFC audio stream and re-encodes it as pcm_s16be — 16-bit signed PCM with big-endian byte order — which is the standard and most compatible codec for AU files. If the source AIFC already uses pcm_s16be, the audio data is essentially transcoded at the same bit depth with minimal quality impact. If the source uses a higher-resolution codec like pcm_s24be or pcm_s32be, the bit depth is reduced to 16-bit, which involves a small, usually inaudible quality trade-off. Compressed AIFC sources (A-law, μ-law) are fully decoded and then stored as linear PCM in the output AU file. The AU format uses a simple fixed header, so AIFC metadata such as embedded markers or application-specific chunks is discarded during the conversion.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg command-line tool, which is the underlying engine powering this browser-based converter via its WebAssembly (FFmpeg.wasm) build.
-i input.aifc Specifies the input file — an AIFC audio file, which may contain uncompressed PCM or compressed audio data (A-law, μ-law, or various PCM bit depths) wrapped in Apple's extended AIFF container.
-c:a pcm_s16be Sets the audio codec for the output to 16-bit signed PCM with big-endian byte order, which is the standard and most universally compatible audio encoding for the AU format. This matches the native byte order of the Sun AU container and ensures the output plays correctly on Unix systems and Java environments.
output.au Specifies the output filename with the .au extension, which tells FFmpeg to use the Sun AU container format (identified by the magic number 0x2e736e64). The AU container wraps the pcm_s16be audio stream with a minimal header containing sample rate, channel count, and encoding type.

Common Use Cases

  • Preparing audio files for playback on legacy Unix or Solaris workstations that natively support the AU format but not AIFC.
  • Converting compressed AIFC audio (A-law or μ-law encoded) to a straightforward uncompressed AU file for use in older Unix-based audio pipelines.
  • Supplying audio assets to Java applications or applets, which have historically used the AU format as a default supported audio type via the javax.sound API.
  • Archiving or migrating Apple-originated AIFC audio from classic Mac OS environments into a format readable by Unix audio tools like 'play' or 'sox'.
  • Stripping AIFC-specific metadata and compression wrappers to produce a bare-bones PCM AU file for use in embedded systems or telephony testing environments.
  • Reproducing early internet audio workflows where AU was the dominant streaming audio format for web browsers in the 1990s, using original AIFC source material.

Frequently Asked Questions

It depends on the source codec. If your AIFC file uses pcm_s16be (16-bit PCM), the conversion is essentially lossless — the audio data is re-wrapped with minimal change. If the source uses higher-resolution codecs like pcm_s24be or pcm_s32be, the bit depth is reduced to 16-bit in the AU output, which can introduce very subtle quantization differences but is generally imperceptible for most listening purposes. If the source is compressed A-law or μ-law, it is fully decoded to linear PCM, which recovers only the quality that the original lossy compression preserved.
The AU (Sun .au) format was designed in the late 1980s as a simple, minimal audio container for Unix systems, and its specification defines only a small set of supported encoding types. Unlike AIFC, which was designed by Apple specifically to support professional audio with extended codec options, AU prioritizes simplicity and portability. Its native codec support tops out at 16-bit signed PCM (pcm_s16be), 8-bit variants, and telephony codecs like A-law and μ-law, reflecting its origins as a workstation and telephony format rather than a professional audio production format.
No. The AU format has an extremely minimal header structure that stores only sample rate, channel count, encoding type, and an optional short annotation string. AIFC-specific metadata — including instrument loop points, markers, MIDI information, and application-specific chunks common in classic Mac OS audio workflows — is not representable in the AU format and will be dropped during conversion. If preserving this metadata is important, you should archive the original AIFC file alongside the converted AU output.
You can replace '-c:a pcm_s16be' with '-c:a pcm_alaw' or '-c:a pcm_mulaw' in the command, as both of these codecs are supported by the AU format. For example: 'ffmpeg -i input.aifc -c:a pcm_mulaw output.au'. This is particularly useful for telephony applications where μ-law (G.711) encoding is a standard requirement, and it will produce a smaller file than 16-bit PCM at the cost of reduced dynamic range.
The single-file command shown (ffmpeg -i input.aifc -c:a pcm_s16be output.au) processes one file at a time. For batch processing on the command line, you can use a shell loop — for example, on Linux or macOS: 'for f in *.aifc; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.aifc}.au"; done'. On Windows Command Prompt: 'for %f in (*.aifc) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. The browser-based tool on this page handles one file at a time, but the desktop FFmpeg command is ideal for bulk conversions of large collections.
Both AIFC (derived from Apple's AIFF) and Sun AU were independently developed on big-endian hardware architectures — Motorola 68k-based Macs and SPARC-based Sun workstations respectively — so both naturally store multi-byte audio samples in big-endian (most significant byte first) order. When converting pcm_s16be AIFC to pcm_s16be AU, FFmpeg does not need to swap byte order in the audio samples themselves; the primary work is replacing the container header. This is a minor efficiency point, though FFmpeg handles byte-order differences transparently regardless of direction.

Technical Notes

The AU format's simplicity is both its strength and its limitation in this conversion. Its fixed-size or variable annotation header contains no facility for bit depths above 16-bit linear PCM, so any AIFC source with 24-bit or 32-bit PCM content (pcm_s24be, pcm_s32be, pcm_f32be, pcm_f64be) will be downsampled to 16-bit on output, reducing the dynamic range ceiling from 144 dB (32-bit float) or 96 dB (16-bit) accordingly. For professional audio archival purposes, this makes AU a poor destination if your AIFC source is high-resolution. However, for telephony, legacy Unix compatibility, or Java audio playback, pcm_s16be AU output is entirely appropriate. The AU format does support multi-channel audio in theory, but many legacy implementations assume mono or stereo, so stereo AIFC files should convert without issues. Channel count above two may cause compatibility problems with older AU players. AU files also lack any concept of chapters, embedded artwork, or subtitle tracks, none of which AIFC supports either, so no rich metadata is lost beyond AIFC-specific audio markers. FFmpeg's AU muxer correctly writes the standard Sun magic number (0x2e736e64) and encoding type field, ensuring the output files are recognized by conforming AU players on Unix systems.

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