MyMediaTools

Convert AU to FLAC — Free Online Tool

Convert Sun AU audio files to FLAC, preserving every bit of the original PCM audio data in a modern lossless compressed container. This tool re-encodes the raw PCM stream (typically pcm_s16be) found in AU files into FLAC, reducing file size without discarding any audio information.

FFmpeg Command

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Estimated output:

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How It Works

AU files store audio as raw PCM data with a minimal header — typically 16-bit big-endian signed PCM (pcm_s16be), though other encodings like A-law and μ-law are also common. During conversion, FFmpeg reads the PCM samples directly from the AU container and feeds them into the FLAC encoder, which applies lossless Rice coding compression. Because FLAC is lossless, every audio sample from the original AU file is reproduced exactly in the output — no quality is sacrificed. The big-endian byte order of AU's native pcm_s16be encoding is transparently handled during decoding, so the FLAC output contains perfectly preserved audio regardless of the source platform (typically Unix/Solaris systems). The resulting FLAC file is typically 40–60% smaller than the raw PCM stored in the AU container.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg tool, which is running here as a WebAssembly binary (FFmpeg.wasm) entirely within your browser — no audio data is sent to any server.
-i input.au Specifies the input AU file. FFmpeg reads the AU header to determine the audio encoding (most commonly pcm_s16be, big-endian 16-bit PCM) and decodes the raw audio samples for processing.
-c:a flac Instructs FFmpeg to encode the decoded PCM audio samples using the FLAC lossless codec. The PCM data from the AU file is compressed using FLAC's linear predictive coding and Rice entropy coding, producing a losslessly compressed output.
-compression_level 5 Sets the FLAC compression effort to level 5 out of a maximum of 8. This is the FLAC default and provides a good balance between encoding speed and output file size. The audio quality is identical at every level — only the degree of compression and encoding time change.
output.flac Defines the output filename. The .flac extension tells FFmpeg to use the FLAC container format, which wraps the encoded FLAC audio stream along with support for Vorbis comment metadata tags and a seekable index — features absent from the original AU file.

Common Use Cases

  • Archiving legacy Unix workstation sound effects or system audio samples originally distributed in AU format for long-term preservation in a modern, widely supported lossless format
  • Preparing early internet audio clips (AU was a dominant web audio format in the mid-1990s) for use in modern audio editors like Audacity or Adobe Audition, which have stronger FLAC support than AU
  • Converting Sun Microsystems Java audio samples (AU was Java's default audio format) into FLAC for use in contemporary multimedia projects or sample libraries
  • Migrating collections of scientific or research audio recordings stored on Unix systems in AU format into FLAC for broader compatibility and indexed metadata tagging
  • Converting μ-law or A-law encoded AU telephony audio to FLAC for lossless archiving before any further processing or analysis
  • Bringing AU audio assets from retro software, games, or operating systems into a format compatible with modern music players, streaming tools, and audio hardware

Frequently Asked Questions

Yes — FLAC is a lossless format, so every PCM sample from the AU file is encoded into FLAC without any modification. If the source AU file uses pcm_s16be (16-bit big-endian PCM), the decoded audio is bit-for-bit identical to the original. Even AU files using compressed encodings like μ-law or A-law are decoded to PCM first and then losslessly re-encoded into FLAC, so the FLAC file faithfully represents what the AU file sounded like.
AU files store audio as raw, uncompressed PCM data — each sample takes a fixed amount of space with no attempt to reduce redundancy. FLAC applies lossless compression using a technique called linear predictive coding followed by Rice entropy coding, which exploits patterns and redundancy in audio waveforms to shrink the data. Typical FLAC files are 40–60% smaller than the equivalent raw PCM, and the compression is entirely reversible — the original samples can be reconstructed perfectly.
AU files have an extremely minimal header that can include a short ASCII text annotation field, but in practice most AU files carry little to no metadata. FLAC, by contrast, supports rich Vorbis comment tags (artist, title, album, etc.). Any annotation text present in the AU header may be carried over, but you will likely want to add metadata manually after conversion using a tag editor like MusicBrainz Picard or Kid3, since the AU source typically lacks structured tag data.
FFmpeg automatically detects and decodes μ-law (pcm_mulaw) and A-law (pcm_alaw) AU files before passing the audio to the FLAC encoder. These are logarithmic compression schemes commonly used in telephony. The decoded audio is typically 16-bit PCM, and FLAC then losslessly compresses that decoded PCM. The resulting FLAC is a lossless representation of the μ-law or A-law decoded audio — note that the original logarithmic quantization from the telephony encoding is preserved in the decoded samples, not reversed.
The compression level is controlled by the -compression_level flag, which accepts values from 0 to 8. The default used here is 5, which balances encoding speed and file size well for most use cases. Changing it to -compression_level 8 will produce the smallest possible FLAC file but encode more slowly, while -compression_level 0 encodes very quickly with minimal compression. Crucially, all levels are lossless — the audio quality is identical regardless of the value you choose, only encoding speed and output file size differ.
Yes. On Linux or macOS, you can loop over all AU files in a directory with a shell command like: for f in *.au; do ffmpeg -i "$f" -c:a flac -compression_level 5 "${f%.au}.flac"; done. On Windows Command Prompt, use: for %f in (*.au) do ffmpeg -i "%f" -c:a flac -compression_level 5 "%~nf.flac". Each AU file will be independently converted to a FLAC file with the same base name.

Technical Notes

AU (Sun Audio) files use big-endian byte ordering, which reflects their Unix/Solaris heritage. FFmpeg's AU demuxer handles the byte-order conversion transparently, so pcm_s16be samples are correctly decoded to native-endian PCM before reaching the FLAC encoder. The AU format supports several audio encodings: straight PCM (pcm_s16be, pcm_s8, pcm_u8), and companded formats (pcm_alaw, pcm_mulaw). All of these are handled by this conversion, though the 8-bit and companded formats will result in lower fidelity in the decoded audio that the FLAC then losslessly preserves. AU has no support for metadata tags, chapters, multiple audio tracks, or subtitles, and FLAC likewise does not support video or subtitles, so this conversion is purely an audio container and codec upgrade. One practical consideration: very old AU files from early Unix systems or Java applications sometimes have non-standard or malformed headers; FFmpeg is generally robust to these but occasional files may require the -f au flag to force AU demuxing. The FLAC output will be fully seekable and compatible with virtually all modern audio software, unlike AU which has limited support outside Unix-oriented and legacy tools.

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