Convert AU to AC3 — Free Online Tool
Convert Sun AU audio files to Dolby Digital AC3 format, transcoding raw PCM audio (typically 16-bit big-endian) into compressed AC3 at 192kbps by default. Ideal for taking legacy Unix audio content and preparing it for DVD authoring, broadcast workflows, or Dolby Digital-compatible playback systems.
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DownloadHow It Works
AU files store uncompressed PCM audio (most commonly pcm_s16be — 16-bit signed big-endian samples) with a minimal header, which means the source audio is raw and lossless. During this conversion, FFmpeg decodes those raw PCM samples and re-encodes them using the AC3 (Dolby Digital) codec, a lossy perceptual compression algorithm that analyzes frequency masking to discard audio information the human ear is unlikely to notice. The result is a significantly smaller .ac3 file with compressed audio at the chosen bitrate. Because AU supports only mono or stereo audio, the output AC3 will preserve that channel layout — it will not artificially upmix to 5.1 surround. This is a full transcode (decode then re-encode), not a remux, so some quality reduction from the original lossless PCM source is inherent.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles decoding the AU PCM audio and re-encoding it as Dolby Digital AC3. |
-i input.au
|
Specifies the input Sun AU file. FFmpeg reads the AU header to determine the PCM encoding type (e.g., pcm_s16be), sample rate, and channel count before decoding the raw audio samples. |
-c:a ac3
|
Selects the AC3 (Dolby Digital) encoder for the audio stream, replacing the lossless PCM data from the AU source with perceptually compressed Dolby Digital audio suitable for DVD, broadcast, and Dolby-compatible devices. |
-b:a 192k
|
Sets the AC3 audio bitrate to 192 kilobits per second, the standard default for stereo Dolby Digital encoding. This balances file size reduction against audio quality when compressing from the original uncompressed PCM source. |
output.ac3
|
Defines the output filename with the .ac3 extension, producing a raw Dolby Digital elementary stream file that can be played directly in supporting software or muxed into DVD/Blu-ray containers for disc authoring. |
Common Use Cases
- Preparing archived Unix workstation audio recordings for inclusion in a DVD video project, where AC3 is a required audio format
- Converting legacy Sun Microsystems system sounds or early internet audio files (.au) into a format accepted by Dolby Digital broadcast encoding pipelines
- Reducing the file size of large uncompressed AU audio files for storage or distribution while maintaining compatibility with set-top boxes and media players that support AC3
- Migrating old scientific or academic audio data recorded in AU format on Unix systems into a compressed format suitable for modern media servers
- Encoding AU-format audio assets from retro software or games into AC3 for use in video production or multimedia presentations requiring Dolby Digital audio tracks
- Creating AC3 test streams from known PCM sources (AU files) for validating Dolby Digital decoder implementations or broadcast equipment
Frequently Asked Questions
Yes, there will be some quality reduction because AC3 is a lossy codec and the AU source is uncompressed PCM. However, at the default 192kbps bitrate, the loss is generally imperceptible for speech and most music content. If the AU file contains high-fidelity audio or critical listening material, increasing the bitrate to 320k or 384k will better preserve the original PCM quality.
No. AU files support only mono or stereo audio due to the format's simple structure and codec limitations. FFmpeg will encode the AC3 output with the same channel layout as the source AU file — typically mono or stereo. AC3 does support up to 5.1 surround, but that capability cannot be used here without a multi-channel source.
Replace the '-b:a 192k' value in the command with your desired bitrate. AC3 supports specific bitrate options: 96k, 128k, 192k, 256k, 320k, 384k, 448k, and 640k. For example, to encode at 384kbps run: ffmpeg -i input.au -c:a ac3 -b:a 384k output.ac3. Higher bitrates produce better audio quality at the cost of larger file size.
pcm_s16be means the AU file stores 16-bit signed integer audio samples in big-endian byte order, which was standard on Sun SPARC and other big-endian Unix architectures. This is fully uncompressed audio, so it is the highest-quality input you can have. FFmpeg will correctly decode these big-endian samples before feeding them to the AC3 encoder, so the byte order difference is handled transparently and has no negative effect on the output.
A standalone .ac3 file is the raw Dolby Digital elementary stream and is not directly playable by most consumer disc players on its own — it needs to be muxed into a container like VOB (for DVD) or M2TS (for Blu-ray) using a DVD or Blu-ray authoring tool. However, the AC3 audio stream produced by this conversion is fully standards-compliant Dolby Digital and will work correctly once multiplexed into the appropriate container.
Yes. On Linux or macOS you can use a shell loop: for f in *.au; do ffmpeg -i "$f" -c:a ac3 -b:a 192k "${f%.au}.ac3"; done. On Windows Command Prompt use: for %f in (*.au) do ffmpeg -i "%f" -c:a ac3 -b:a 192k "%~nf.ac3". This processes every AU file in the current directory and outputs a matching AC3 file, which is especially useful for migrating large archives of legacy Unix audio content.
Technical Notes
AU (Sun Audio) files use a straightforward header containing sample rate, channel count, encoding type, and an optional annotation field, followed by raw audio data. The most common encoding found in AU files is pcm_s16be (16-bit big-endian PCM), though pcm_s8, pcm_u8, pcm_alaw, and pcm_mulaw variants also exist — FFmpeg auto-detects the encoding from the header and handles each correctly before passing decoded samples to the AC3 encoder. AC3 uses a transform-based perceptual coder operating on 512-sample blocks with dynamic bit allocation, which means the bitrate is distributed across frequency bands based on masking thresholds. Because AU does not store metadata beyond a plain-text annotation field, no meaningful metadata (artist, title, album) is carried through to the AC3 output. The AC3 format itself also has very limited metadata support, so this is not a significant loss. One important limitation: AC3 requires a sample rate of 32000, 44100, or 48000 Hz. If your AU source uses an unusual sample rate (some Unix systems recorded at 8000 Hz for telephone audio), FFmpeg will automatically resample to the nearest compatible rate, which may subtly affect the character of the audio. Files containing mulaw or alaw encoded AU audio (common in early VoIP and telephony recordings) will be decoded correctly but will reveal the pre-existing bandwidth limitations of those encodings in the AC3 output.