Convert AC3 to CAF — Free Online Tool
Convert AC3 (Dolby Digital) audio files to Apple's Core Audio Format (CAF), decoding the lossy surround sound stream and re-encoding it as uncompressed PCM for maximum compatibility within the Apple ecosystem. This is especially useful when you need to bring Dolby Digital broadcast or disc audio into macOS or iOS audio workflows.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AC3 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
AC3 is a lossy compressed audio format that often carries multi-channel surround sound (5.1, etc.) encoded with Dolby Digital. CAF does not natively support the AC3 codec, so FFmpeg fully decodes the AC3 bitstream — including any surround channels — and re-encodes the audio as 16-bit signed little-endian PCM (pcm_s16le) inside the CAF container. This means the compressed Dolby Digital data is decompressed to raw waveform data, which is lossless from this point forward but does carry forward any quality loss already introduced by the original AC3 encoding. The resulting CAF file is uncompressed, so it will be significantly larger than the source AC3 file but fully compatible with Core Audio on macOS and iOS.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, which handles all decoding, codec processing, and container writing for this AC3-to-CAF conversion entirely within your browser via WebAssembly. |
-i input.ac3
|
Specifies the input file — an AC3 Dolby Digital audio file. FFmpeg reads and decodes this compressed bitstream, which may contain stereo or multi-channel surround audio, before passing the raw audio to the output encoder. |
-c:a pcm_s16le
|
Sets the output audio codec to 16-bit signed little-endian PCM, the uncompressed audio format native to Core Audio and compatible with all Apple platforms. This replaces the lossy Dolby Digital compression from the source with lossless, sample-accurate storage inside the CAF container. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps, but this flag has no practical effect on PCM codecs like pcm_s16le — the actual bitrate of uncompressed PCM is determined solely by sample rate, bit depth, and channel count. It is included as a default parameter and is safely ignored by FFmpeg. |
output.caf
|
Defines the output filename and tells FFmpeg to wrap the PCM audio stream in Apple's Core Audio Format container (.caf), which supports unlimited file sizes and is natively readable by macOS, iOS, and Apple audio frameworks like AVFoundation and Core Audio. |
Common Use Cases
- Importing Dolby Digital audio ripped from a DVD or Blu-ray into Logic Pro or GarageBand for music production or sound design work
- Preparing broadcast television AC3 audio tracks for use in Final Cut Pro or other Apple-native editing tools that prefer CAF or PCM audio
- Archiving AC3 surround sound stems from a video project as uncompressed CAF files for long-term preservation in an Apple-centric post-production pipeline
- Converting AC3 audio extracted from a video file into CAF so it can be loaded into an iOS app or Swift-based audio processing project using AVFoundation
- Decoding a Dolby Digital 5.1 track from a broadcast recording to uncompressed PCM in CAF format before downmixing or re-encoding for a stereo deliverable
Frequently Asked Questions
No — converting AC3 to CAF with PCM encoding will not recover quality lost during the original AC3 compression. AC3 is a lossy format, meaning some audio data was permanently discarded when the file was first encoded as Dolby Digital. The CAF output will be an uncompressed representation of what remains after that lossy step, so it is lossless going forward but cannot undo prior quality loss. Think of it as freezing the current quality in an uncompressed form.
FFmpeg decodes all channels present in the AC3 file and writes them into the PCM stream inside the CAF container. If your AC3 source is a 5.1 file, the output CAF will contain six channels of uncompressed PCM audio. Core Audio on macOS supports multi-channel PCM in CAF files, so the surround information should be preserved and accessible to compatible software like Logic Pro.
AC3 uses Dolby Digital compression, which typically reduces audio data to a fraction of its raw size — a 192 kbps AC3 file stores far less data per second than uncompressed audio. When you convert to CAF with pcm_s16le, every audio sample is stored as a full 16-bit integer with no compression applied. For CD-quality stereo audio this means roughly 10 MB per minute, and for 5.1 surround the size multiplies by the number of channels. This is expected behavior for uncompressed PCM output.
Yes. CAF supports multiple PCM variants, and you can change the codec flag to use a higher bit depth. Replace '-c:a pcm_s16le' with '-c:a pcm_s24le' for 24-bit audio, which some professional macOS applications prefer. Note that the '-b:a' bitrate flag is ignored for PCM codecs since the bitrate is determined entirely by sample rate, bit depth, and channel count — you can safely remove it when switching to PCM variants.
For PCM codecs like pcm_s16le, the '-b:a' bitrate flag has no effect — PCM audio is uncompressed and its bitrate is fixed by the sample rate, bit depth, and channel count rather than a target bitrate setting. The flag is included in the command as a default parameter but is effectively ignored by FFmpeg in this context. If you run the command locally you will see FFmpeg disregard it without throwing an error.
On macOS or Linux you can use a shell loop: 'for f in *.ac3; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.ac3}.caf"; done'. On Windows Command Prompt use: 'for %f in (*.ac3) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.caf"'. This processes every AC3 file in the current directory and outputs a matching CAF file with the same base filename.
Technical Notes
AC3 audio commonly carries between 1 and 5.1 channels at bitrates ranging from 96 kbps to 640 kbps, and the decoder FFmpeg uses (the native ac3 decoder) handles standard Dolby Digital streams reliably. The CAF container was designed by Apple to overcome the 4 GB file size limit of AIFF and WAV, making it well-suited for long multi-channel recordings. The default output codec here is pcm_s16le (signed 16-bit little-endian PCM), which provides CD-quality resolution and broad compatibility with Core Audio APIs. If the source AC3 has a sample rate other than 44.1 kHz or 48 kHz, FFmpeg will preserve the original sample rate in the CAF output unless you explicitly resample with '-ar'. Metadata embedded in AC3 streams (such as dialnorm and dynamic range metadata specific to Dolby Digital) is format-specific and will not be carried over to the PCM CAF file, as CAF uses its own metadata chunk structure. CAF files are not widely supported outside the Apple ecosystem, so this conversion is best suited for workflows that terminate in macOS or iOS environments.