Convert AU to MP3 — Free Online Tool
Convert Sun AU audio files to MP3 by transcoding raw PCM audio (typically 16-bit big-endian) through the LAME encoder, producing a compressed, universally playable MP3 file. Ideal for modernizing legacy Unix audio recordings that most modern players and devices cannot open.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
AU files store uncompressed PCM audio (most commonly 16-bit signed big-endian, as defined in the file's simple header) with no lossy compression. During conversion, FFmpeg reads the raw PCM samples directly from the AU container, then passes them through the LAME MP3 encoder (libmp3lame), which applies psychoacoustic modeling to discard audio information that human hearing is least sensitive to, compressing the data significantly. Because AU stores uncompressed audio, the source material is pristine — no generation loss is introduced from the input side. The output MP3 uses a default bitrate of 128 kbps, which typically reduces file size by 80–90% compared to 16-bit PCM AU at standard sample rates, while preserving perceptually acceptable audio quality.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. This is the same underlying engine running in your browser via WebAssembly (FFmpeg.wasm) — the command shown is fully portable to any desktop FFmpeg installation. |
-i input.au
|
Specifies the input AU file. FFmpeg reads the AU header to detect the PCM encoding variant (e.g., 16-bit big-endian, µ-law, or A-law), sample rate, and channel count before decoding the raw audio data. |
-c:a libmp3lame
|
Selects the LAME MP3 encoder for the audio stream, transcoding the raw PCM samples from the AU file into compressed MP3 audio using psychoacoustic modeling to reduce file size. |
-b:a 128k
|
Sets the MP3 output bitrate to 128 kilobits per second, a widely used default that balances file size and audio quality. Increase to 192k or 320k for higher fidelity, or reduce to 64k for smaller files when quality is less critical. |
output.mp3
|
Defines the output filename and container. The .mp3 extension tells FFmpeg to write an MP3 bitstream file with ID3 tag support, compatible with virtually all modern media players, browsers, and mobile devices. |
Common Use Cases
- Playing back legacy Unix workstation recordings or system sounds (.au files) on modern smartphones, media players, and streaming apps that only support MP3.
- Archiving old Sun Microsystems or NeXT audio recordings from the 1990s into a more durable, widely supported format before the original playback software becomes unavailable.
- Converting AU-format audio samples downloaded from early internet sound archives (common in the late 1990s web era) into MP3 for use in modern DAWs or sample libraries.
- Reducing the file size of large uncompressed AU recordings for email attachment, web hosting, or mobile sharing without purchasing additional software.
- Batch-converting AU sound effect libraries from vintage Unix or Solaris systems into MP3 for use in contemporary video editing or game development pipelines.
- Extracting and compressing telephone-quality AU files (often encoded with µ-law or A-law PCM) into MP3 for podcast archiving or audio documentation workflows.
Frequently Asked Questions
Yes — MP3 is a lossy format, so some audio information is permanently discarded during encoding. However, since AU files store uncompressed PCM audio, the source is as clean as possible before encoding begins, meaning no compounded quality loss from a previous lossy stage. At the default 128 kbps bitrate, the result is perceptually acceptable for most speech and music. If the AU file contains µ-law or A-law encoded audio (telephone-quality, 8-bit), the quality ceiling of the source itself is already limited, so the MP3 output will reflect that, not introduce additional degradation.
The AU format is a legacy Unix audio container developed by Sun Microsystems in the 1980s and 1990s. While technically simple and once common on early internet pages, native AU playback support was dropped from most consumer software and operating systems over the past two decades. Converting to MP3 solves this immediately, as MP3 is supported by virtually every media player, browser, smartphone, and streaming platform in use today.
Replace the '128k' value in the '-b:a 128k' flag with your desired bitrate. Common options are 64k (small file, lower quality, good for speech), 192k (higher quality, larger file), or 320k (maximum standard MP3 quality). For example: 'ffmpeg -i input.au -c:a libmp3lame -b:a 192k output.mp3'. Higher bitrates produce larger files but better audio fidelity, which matters most for music with complex high-frequency content.
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.au; do ffmpeg -i "$f" -c:a libmp3lame -b:a 128k "${f%.au}.mp3"; done'. On Windows Command Prompt: 'for %f in (*.au) do ffmpeg -i "%f" -c:a libmp3lame -b:a 128k "%~nf.mp3"'. This processes each AU file in the current directory and outputs a corresponding MP3 with the same base filename.
AU files have a very minimal header that can optionally contain a plain-text annotation field, but this is rarely populated in practice and does not map to standard MP3 ID3 tags. FFmpeg will attempt to carry over any annotation data, but the resulting MP3 will generally have no embedded artist, title, or album metadata. You can add ID3 tags manually using the '-metadata' flag — for example: 'ffmpeg -i input.au -c:a libmp3lame -b:a 128k -metadata title="My Recording" output.mp3'.
FFmpeg handles this automatically. The AU format supports multiple PCM variants including pcm_mulaw (µ-law) and pcm_alaw (A-law), both of which are 8-bit logarithmic encodings historically used for telephony. FFmpeg detects the encoding from the AU file header and decodes it to linear PCM internally before handing the samples to the LAME encoder. No changes to the FFmpeg command are needed — the conversion command works identically regardless of which PCM variant your AU file uses.
Technical Notes
AU files use a straightforward 24-byte minimum header storing the magic number, data offset, data size, encoding type, sample rate, and channel count — making them easy for FFmpeg to parse reliably. The most common AU encoding is 16-bit signed big-endian PCM (pcm_s16be), though 8-bit µ-law (pcm_mulaw) and A-law (pcm_alaw) variants are also frequently encountered, especially in files originating from telephony or early Unix systems. All of these decode to linear PCM before LAME encoding, so the transcoding path is clean. The output MP3 is encoded using libmp3lame, the most mature and widely tested open-source LAME implementation, which uses joint stereo encoding for stereo sources and handles mono AU files correctly by encoding a mono MP3 stream. AU files have no concept of chapters, subtitles, or multiple audio tracks, so none of those are lost in conversion. File size reduction is dramatic for standard-quality sources: a 10-minute mono AU file at 44.1 kHz 16-bit PCM (~52 MB) will compress to roughly 9 MB at 128 kbps MP3. Note that MP3 adds a small amount of encoder delay (typically 576 samples of padding) at the beginning of the file; this is standard behavior and is usually handled transparently by compliant MP3 decoders.