Convert AC3 to AU — Free Online Tool
Convert AC3 (Dolby Digital) audio files to Sun AU format, transcoding the compressed surround-sound bitstream into uncompressed PCM audio (16-bit big-endian) compatible with Unix systems and legacy audio toolchains. Ideal for bringing broadcast-quality AC3 audio into environments that expect raw PCM data without proprietary codecs.
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FFmpeg Command
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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
AC3 is a lossy compressed format that encodes up to 5.1 channels of surround audio using perceptual coding to discard frequencies the human ear is less sensitive to. The Sun AU format, by contrast, stores raw uncompressed PCM audio — in this case 16-bit signed big-endian samples (pcm_s16be). During conversion, FFmpeg decodes the AC3 bitstream fully, reconstructing the PCM waveform, then re-encodes it into AU's simple header structure followed by raw audio data. If the source is 5.1 surround, FFmpeg will downmix the channels to stereo (or whatever the output channel count supports) since AU is a simple stereo/mono container. The result is a larger but fully decompressed audio file that requires no special codec to play back — any software that can read raw PCM or AU files can handle it.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles the AC3 decoding and PCM encoding pipeline entirely within your browser via WebAssembly — no server is involved. |
-i input.ac3
|
Specifies the input Dolby Digital AC3 file. FFmpeg reads the compressed AC3 bitstream and prepares to decode it into raw PCM audio samples for re-encoding. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, the standard uncompressed encoding used by the Sun AU format. This replaces the perceptually compressed AC3 bitstream with raw, decoder-ready audio samples in the byte order AU expects. |
output.au
|
Defines the output file name and triggers FFmpeg to wrap the pcm_s16be audio stream in a Sun AU container, writing the simple AU header followed by the raw PCM audio data. |
Common Use Cases
- Extracting clean, uncompressed audio from a Dolby Digital AC3 track ripped from a DVD for use in audio editing tools on Unix or Linux workstations that natively support AU files.
- Feeding AC3 broadcast audio into legacy Unix-based signal processing pipelines (e.g., SoX, early Java sound APIs) that expect AU-formatted PCM input rather than compressed audio streams.
- Converting AC3 audio captured from broadcast television into a format suitable for archival or waveform analysis tools that require raw uncompressed samples.
- Preparing Dolby Digital audio for scientific or acoustic research workflows running on Unix systems where AU is the standard audio interchange format.
- Stripping the Dolby Digital compression from a movie's audio track to produce a lossless-quality reference file for A/B comparison against other codec outputs.
- Enabling playback of AC3 surround content on legacy Sun Microsystems workstations or software environments that only support the AU format natively.
Frequently Asked Questions
Not in any additional meaningful way. AC3 is a lossy format, so the perceptual quality loss was introduced when the AC3 file was originally created. During this conversion, FFmpeg decodes the AC3 bitstream to its best PCM representation and stores that exactly in the AU file as uncompressed 16-bit samples. No further lossy compression is applied. The AU output faithfully captures the decoded AC3 audio — you won't degrade the signal further, but you also cannot recover the quality lost during the original AC3 encoding.
AU is a simple audio container without explicit multi-channel surround support in the way modern formats like MKV or MP4 handle it. FFmpeg will typically downmix the AC3's surround channels (center, LFE, rear surrounds) into a stereo or mono PCM stream during the conversion. If preserving the full surround channel layout is important, AU is not an appropriate target format — consider using WAV or AIFF instead, which can store multi-channel PCM properly.
AC3 achieves file sizes roughly 10–20x smaller than uncompressed audio by using perceptual lossy compression. The Sun AU format stores raw, uncompressed PCM samples — every audio sample is written directly to disk with no compression. A typical 192 kbps AC3 file might expand to several times its size when converted to 16-bit PCM in AU format. This is expected behavior and is the trade-off for universal readability without a decoder.
No. The Sun AU format has an extremely minimal header — it stores only the encoding type, sample rate, channel count, and an optional brief annotation field. It has no support for structured metadata tags, chapter markers, or track information. Any metadata embedded in the AC3 source file will not be carried over to the AU output. If metadata preservation matters, consider converting to FLAC or WAV, which have robust tagging support.
Replace '-c:a pcm_s16be' in the command with your preferred codec. The AU format supports several PCM variants: 'pcm_mulaw' for G.711 mu-law (common in telephony), 'pcm_alaw' for A-law, 'pcm_s8' for 8-bit signed PCM, or 'pcm_u8' for 8-bit unsigned PCM. For example: 'ffmpeg -i input.ac3 -c:a pcm_mulaw output.au'. Note that mu-law and A-law are lossy telephone-quality codecs and will noticeably reduce audio fidelity compared to pcm_s16be.
Yes, with a simple shell loop. On Linux or macOS, run: 'for f in *.ac3; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.ac3}.au"; done'. On Windows Command Prompt, use: 'for %f in (*.ac3) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.au"'. Each AC3 file will be decoded and written as a separate AU file. Keep in mind that batch conversion of multiple AC3 files may produce very large AU outputs due to the uncompressed nature of the format.
Technical Notes
The Sun AU format uses a big-endian byte order for its PCM data, which is why the default codec for this conversion is pcm_s16be (16-bit signed big-endian) rather than the little-endian pcm_s16le used by WAV. This big-endian convention originates from the SPARC/Motorola architectures that Sun workstations used. AC3 supports sample rates up to 48 kHz, and FFmpeg will preserve the source sample rate in the AU output — no resampling occurs unless explicitly requested. The AU format technically allows extensible annotation headers but virtually no modern software writes meaningful data there. One important limitation: AC3 audio using a 5.1 channel layout will be downmixed during decoding if the output channel configuration is not explicitly set, so users who need to verify surround channel integrity should inspect the decoded output. Since AU stores raw samples, the file is immediately suitable for importing into waveform editors or analysis tools that support reading raw audio data, making it useful as an intermediate format even if AU is not the final destination.