Convert AC3 to AIFF — Free Online Tool
Convert AC3 (Dolby Digital) audio files to AIFF, decoding the lossy surround sound compression into uncompressed PCM audio stored in Apple's lossless container. Ideal for bringing broadcast or DVD audio into professional macOS audio workflows where uncompressed fidelity is required.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AC3 file here
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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 codec — it permanently discards audio data during encoding using perceptual compression, which is why Dolby Digital files are compact enough for DVD and broadcast use. Converting to AIFF does not recover that lost data; instead, FFmpeg decodes the AC3 bitstream into raw PCM samples and writes them into the AIFF container using the pcm_s16be codec (16-bit, big-endian, signed PCM — the native format Apple designed AIFF around). The result is an uncompressed file where every decoded sample is stored verbatim, with no further generation loss introduced by the conversion itself. If the source AC3 carries 5.1 surround channels, all six channels are preserved in the output AIFF file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles decoding the AC3 input and encoding the PCM output into an AIFF container entirely in your browser via WebAssembly. |
-i input.ac3
|
Specifies the input file — an AC3 Dolby Digital audio stream. FFmpeg automatically detects the AC3 codec and prepares to decode its compressed audio blocks into raw PCM samples. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM, which is the native uncompressed audio encoding for the AIFF format. This decodes the lossy AC3 compression into a flat, uncompressed sample stream compatible with Apple's AIFF container and macOS audio tools. |
output.aiff
|
Defines the output filename and tells FFmpeg to wrap the PCM audio in an AIFF container — Apple's uncompressed audio format. The .aiff extension is recognized natively by macOS and professional DAWs like Logic Pro and Pro Tools. |
Common Use Cases
- Importing Dolby Digital audio extracted from a DVD or Blu-ray rip into Logic Pro, Pro Tools, or another professional macOS DAW that works best with uncompressed AIFF files
- Archiving AC3 broadcast audio captures in an uncompressed format so future re-encoding to any target codec starts from the highest possible decoded quality
- Preparing AC3 dialogue or effects tracks from video production for integration into an Apple-centric post-production pipeline where AIFF is the studio standard
- Expanding AC3 surround sound tracks from a home theater recording or broadcast capture into AIFF for analysis or editing in an audio restoration workflow
- Converting AC3 audio from an old DVD authoring project into AIFF so the stems can be remixed or repurposed without introducing additional codec generations
Frequently Asked Questions
No — because AC3 is a lossy format, the audio data discarded during the original Dolby Digital encoding is gone permanently. The AIFF output will be an uncompressed, bit-perfect representation of the decoded AC3 audio, which means no additional quality loss is introduced by this conversion, but the quality ceiling is set by the original AC3 bitrate. Think of it as freezing the decoded quality in place rather than restoring it.
AC3 uses aggressive perceptual compression — a typical 5.1 AC3 stream at 384 kbps might be 10–20x smaller than its uncompressed equivalent. AIFF stores raw PCM samples with no compression at all, so a 16-bit stereo AIFF at 44.1 kHz uses roughly 10 MB per minute, and a 5.1 surround AIFF uses three times that. This size increase is expected and reflects the removal of compression, not a problem with the conversion.
Yes. FFmpeg decodes all audio channels present in the AC3 stream — whether stereo, 5.1, or another layout — and writes them all into the AIFF output. The pcm_s16be codec supports multi-channel audio, so a 6-channel Dolby Digital source will produce a 6-channel AIFF file. You should verify channel mapping in your DAW if precise surround routing matters for your project.
The pcm_s16be codec (16-bit signed big-endian PCM) is the default for AIFF and matches the bit depth of CD-quality audio, which is sufficient for most AC3 sources since AC3 itself rarely exceeds 16-bit effective resolution after lossy decoding. If you are targeting a professional workflow that requires 24-bit depth — common in post-production — you can change the command to use '-c:a pcm_s24be' for a 24-bit AIFF output. The file size will increase by 50% but the container will be more compatible with high-resolution DAW projects.
You can use a shell loop to batch process files. On macOS or Linux, run: for f in *.ac3; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.ac3}.aiff"; done — this iterates over every .ac3 file in the current directory and produces a matching .aiff file. On Windows Command Prompt, use: for %f in (*.ac3) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". The browser tool processes one file at a time, so the FFmpeg command is especially useful for bulk conversions.
AC3 files typically carry very minimal metadata — they are primarily a raw codec stream designed for embedding in containers like VOB or TS rather than standalone playback. Most standalone AC3 files contain no ID3 or other tag data. AIFF supports metadata via its ANNO and NAME chunks, but if the source AC3 has no tags, the output AIFF will also have none. If metadata preservation is important, tag the AIFF manually after conversion using a tool like mp3tag or directly within your DAW.
Technical Notes
AC3 encodes audio using Dolby's transform-based perceptual coder, which operates on 512-sample blocks and discards frequencies deemed inaudible by a psychoacoustic model. Once decoded by FFmpeg, these samples are written into AIFF as pcm_s16be — 16-bit, big-endian, signed integers, the byte order Apple specified for AIFF when designing it for Motorola 68k processors. The big-endian byte ordering is purely a structural detail of the AIFF container and is handled transparently by any modern macOS application. One practical consideration: if your AC3 source uses dynamic range compression (a common broadcast loudness feature in Dolby Digital), FFmpeg applies it by default during decoding. You can disable it with '-dialnorm 0' or adjust with the 'drc_scale' option if you want the full uncompressed dynamic range. The AIFF format does not support subtitles, chapters, or multiple audio tracks within a single file, but since AC3 is also a single-track audio-only format, no content is lost due to container limitations in this conversion.