Convert AU to WMA — Free Online Tool
Convert Sun AU audio files to Windows Media Audio (WMA) format directly in your browser, transcoding raw PCM audio (typically 16-bit big-endian) into Microsoft's wmav2 codec at 128kbps. This is especially useful for making Unix-originated audio files compatible with Windows Media Player and other Microsoft ecosystem tools.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AU file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
AU files store audio as uncompressed PCM data (most commonly 16-bit signed big-endian, pcm_s16be) with a minimal header — there is no compression to undo, just raw audio samples to read. During conversion, FFmpeg decodes those PCM samples and re-encodes them using the WMA v2 codec (wmav2) at 128kbps, applying lossy compression for the first time. The container changes from the simple Sun AU format (a Unix legacy format with a bare-bones 24-byte header) to the ASF (Advanced Systems Format) container that WMA uses, which adds support for metadata tags. Because WMA is inherently lossy, some audio fidelity is permanently traded away in exchange for a significantly smaller file size.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles all decoding, encoding, and container muxing in this conversion. In the browser version this runs as FFmpeg.wasm compiled to WebAssembly, with no server involvement. |
-i input.au
|
Specifies the input Sun AU file. FFmpeg reads the AU header to determine the PCM codec (e.g., pcm_s16be, pcm_mulaw) and audio parameters such as sample rate and channel count before decoding the raw audio data. |
-c:a wmav2
|
Sets the audio encoder to WMA v2 (wmav2), Microsoft's second-generation Windows Media Audio codec. This is the standard choice for WMA output — it provides better compression efficiency and quality than the older wmav1 and is supported by virtually all WMA-compatible players and devices. |
-b:a 128k
|
Sets the target audio bitrate to 128 kilobits per second for the wmav2 encoder. This is a standard quality level for WMA that balances file size and audio fidelity — approximately 10x smaller than the uncompressed PCM source in a typical AU file at 44.1kHz/16-bit stereo. |
output.wma
|
Defines the output filename with the .wma extension. FFmpeg uses this extension to select the ASF container format, which is the standard container for WMA audio and enables metadata tag support absent from the source AU format. |
Common Use Cases
- Preparing legacy Unix system audio alerts or recordings for playback on Windows machines that expect WMA or Windows Media Player-compatible files
- Reducing the file size of large uncompressed AU recordings (e.g., archived voicemails or early internet audio clips) for storage or email distribution on Windows-centric networks
- Integrating AU audio assets from old Sun workstation software into Windows-based multimedia projects or presentations that require WMA input
- Converting AU files sourced from early web audio archives or NeXT/Sun applications into a format streamable through legacy Windows Media Services infrastructure
- Migrating audio content from Unix/Linux servers to a Windows SharePoint or media library environment where WMA is the standard ingest format
Frequently Asked Questions
Yes, this conversion is lossy. AU files using pcm_s16be store audio as uncompressed PCM — every sample is preserved exactly as recorded. WMA v2 (wmav2) uses perceptual audio coding to discard information the human ear is less likely to notice, achieving compression at the cost of some fidelity. At the default 128kbps setting the result is generally acceptable for voice and typical listening, but the original lossless quality cannot be recovered from the WMA output.
AU files with PCM audio are completely uncompressed, so file size scales directly with duration, sample rate, and bit depth — a 1-minute stereo AU file at 44.1kHz/16-bit is roughly 10MB. WMA at 128kbps compresses that same audio to about 1MB per minute. The size reduction is dramatic precisely because the source format applies zero compression, and wmav2's lossy encoding achieves roughly a 10:1 size ratio at typical settings.
Not reliably. The Sun AU format has an extremely minimal header that can store a small annotation string, but this field is rarely populated in practice and is not a structured metadata system. WMA uses the ASF container, which supports rich metadata tags (artist, title, album, etc.). FFmpeg will attempt to map any annotation data it finds, but in most AU-to-WMA conversions the resulting WMA file will have empty or minimal tags that you would need to fill in separately.
Replace the '128k' value in the '-b:a 128k' flag with your desired bitrate. For example, use '-b:a 192k' for higher quality or '-b:a 96k' for a smaller file. WMA v2 works well at bitrates between 64k and 320k — 128k is a reasonable default for general listening, while 192k or 256k is preferable for music where fidelity matters more. The full command would look like: ffmpeg -i input.au -c:a wmav2 -b:a 192k output.wma
Yes, by replacing '-c:a wmav2' with '-c:a wmav1' in the command. WMA v1 is an older codec with slightly lower compression efficiency and is supported by a narrower range of devices. WMA v2 (wmav2) is the default because it produces better audio quality at the same bitrate and has broader compatibility with Windows Media Player and other software. There is rarely a practical reason to choose WMA v1 unless targeting very old hardware or software that specifically requires it.
The single-file command shown here can be adapted for batch processing in a shell script. On Linux or macOS, you can run: for f in *.au; do ffmpeg -i "$f" -c:a wmav2 -b:a 128k "${f%.au}.wma"; done. On Windows Command Prompt: for %f in (*.au) do ffmpeg -i "%f" -c:a wmav2 -b:a 128k "%~nf.wma". The browser tool processes one file at a time, so the FFmpeg command is particularly valuable for batch jobs involving many AU files.
Technical Notes
The Sun AU format is one of the oldest digital audio formats still encountered, originating from Sun Microsystems and NeXT. Its codec options are limited to raw PCM variants (pcm_s16be being the most common, though pcm_alaw and pcm_mulaw — both telephony-oriented codecs — are also valid). When the AU source uses pcm_alaw or pcm_mulaw, FFmpeg decodes these 8-bit companded formats to linear PCM internally before re-encoding to WMA, so the source codec does not affect the output quality ceiling — what matters is the original recording quality. WMA v2 (wmav2) uses a modified discrete cosine transform (MDCT) and is the same core technology as Windows Media Audio Standard. The ASF container wrapping the WMA stream supports DRM, but FFmpeg does not apply any DRM in this conversion. One known limitation: AU files can in principle contain multichannel audio, but the format's lack of a formal channel layout specification means FFmpeg may default to mono or stereo assumptions; verify channel count before converting if working with unusual source files. File sizes will drop substantially compared to PCM AU sources — expect roughly 8–12x reduction at 128kbps for standard 44.1kHz stereo material.