Convert CAF to AC3 — Free Online Tool
Convert CAF audio files to AC3 (Dolby Digital) format using FFmpeg in your browser — no upload required. This tool re-encodes Apple's Core Audio Format into the AC3 codec favored by DVD, Blu-ray, and broadcast workflows, enabling compatibility with home theater systems and professional A/V equipment.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your CAF 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
CAF files commonly store uncompressed PCM audio (such as pcm_s16le) or losslessly compressed audio, which must be fully decoded and then re-encoded into Dolby Digital's AC3 lossy compression format. Unlike a simple remux, this is a full transcode: the raw audio samples are read from the CAF container, decoded, and then passed through the AC3 encoder at the target bitrate (default 192k). Because AC3 is inherently a lossy format, some audio fidelity is permanently discarded during encoding. The resulting .ac3 file is a raw Dolby Digital bitstream suitable for muxing into DVD VOB files, Blu-ray M2TS containers, or broadcast transport streams.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion both in the browser (via WebAssembly) and on your local desktop installation. |
-i input.caf
|
Specifies the input file in Apple's Core Audio Format. FFmpeg will demux the CAF container and decode whatever audio codec is stored inside it (commonly pcm_s16le, AAC, or FLAC) into raw audio samples for re-encoding. |
-c:a ac3
|
Sets the audio encoder to ac3, which produces Dolby Digital-compliant output. This triggers a full transcode of the decoded CAF audio into AC3's perceptual lossy compression scheme, making the output compatible with DVD players, AV receivers, and broadcast systems. |
-b:a 192k
|
Sets the AC3 audio bitrate to 192 kilobits per second, the default for this tool and a standard choice for stereo Dolby Digital. Increase to 384k or higher if encoding 5.1 surround content, or lower to 128k to reduce file size for mono or voice-only material. |
output.ac3
|
The output filename with the .ac3 extension, which tells FFmpeg to write a raw Dolby Digital elementary bitstream. This file can be played directly in VLC or FFplay, or muxed into a DVD VOB, Blu-ray M2TS, or MKV container for broader compatibility. |
Common Use Cases
- Preparing Apple Logic Pro or GarageBand audio exports (saved as CAF) for inclusion in a DVD or Blu-ray authoring project that requires Dolby Digital AC3 audio tracks.
- Converting CAF recordings from iOS apps into AC3 so they can be played back on AV receivers and home theater systems that decode Dolby Digital.
- Transcoding high-resolution CAF master audio files into AC3 for broadcast television delivery, where AC3 is a mandated or preferred audio format.
- Re-encoding a CAF audio asset from a macOS game or app into AC3 for integration into a video file destined for DVD or set-top box playback.
- Generating an AC3 audio track from CAF source material to test surround sound mixing and Dolby Digital compatibility before a full production encode.
- Archiving Apple-specific CAF audio in a widely supported Dolby Digital format to ensure long-term playback compatibility outside the Apple ecosystem.
Frequently Asked Questions
Yes, always. CAF files often contain uncompressed PCM or losslessly compressed audio, while AC3 is a lossy codec that uses perceptual compression to discard audio data. At the default 192k bitrate, the result is good quality for most listening and broadcast use, but it will never be bit-for-bit identical to the original. If your source CAF contains already-lossy audio (such as AAC), converting to AC3 means going through two generations of lossy encoding, which can introduce compounding artifacts.
AC3 supports up to 5.1 surround sound (six channels: front left, front right, center, low-frequency effects, surround left, surround right). If your CAF source file is stereo or mono, the AC3 output will be stereo or mono respectively — AC3 does not upmix channels. If your source is multichannel (e.g., 5.1 from a Logic Pro project), FFmpeg will encode all channels into the AC3 5.1 layout provided the bitrate is sufficient (384k or higher is recommended for 5.1).
The output is a raw AC3 bitstream file (.ac3), not wrapped in a container like MP4 or MKV. Some media players expect AC3 audio to be muxed inside a container. Most professional players (VLC, FFplay) will play a raw .ac3 file directly, but if you need broader compatibility, consider muxing the AC3 stream into an MP4 or MKV container after conversion. Additionally, some consumer devices only accept AC3 when embedded in a DVD VOB or Blu-ray M2TS stream.
Replace the value after -b:a in the command. For example, use -b:a 384k for higher quality or surround sound, or -b:a 128k to reduce file size for voice-only content. AC3 supports bitrates from 96k up to 640k — for stereo content, 192k to 256k is a practical range, while 5.1 surround typically requires at least 384k to avoid audible artifacts. Note that increasing the bitrate beyond what the source material contains will not recover lost detail.
No. A raw AC3 bitstream file has no metadata container — there is no standardized way to embed title, artist, or other tags in a plain .ac3 file. Any metadata stored in the CAF file (such as recording date or channel layout annotations specific to Apple's format) will be lost during conversion. If metadata preservation is important, mux the AC3 stream into a container like MKV or MP4 that supports both AC3 audio and tagging.
The browser-based tool processes one file at a time, but the displayed FFmpeg command can be adapted for batch processing on your desktop. On Linux or macOS, use a shell loop: for f in *.caf; do ffmpeg -i "$f" -c:a ac3 -b:a 192k "${f%.caf}.ac3"; done. On Windows Command Prompt, use: for %f in (*.caf) do ffmpeg -i "%f" -c:a ac3 -b:a 192k "%~nf.ac3". This is especially useful for large CAF files over 1GB that exceed the browser tool's file size limit.
Technical Notes
CAF was designed by Apple to remove the 4GB file size ceiling of AIFF and WAV, making it common in professional macOS and iOS audio workflows — Logic Pro, GarageBand, and Core Audio APIs all produce CAF natively. The default CAF codec in this tool is pcm_s16le (16-bit signed little-endian PCM), though CAF can also carry AAC, FLAC, Opus, Vorbis, 24-bit, 32-bit, and floating-point PCM. Regardless of which codec is inside the CAF, FFmpeg will fully decode it before re-encoding to AC3. The AC3 encoder in FFmpeg follows the Dolby Digital specification and enforces channel-count-dependent bitrate constraints — for example, mono AC3 has a lower maximum bitrate than 5.1. Sample rate is also relevant: AC3 natively supports 32kHz, 44.1kHz, and 48kHz; if your CAF source uses an unusual sample rate (e.g., 22.05kHz from an iOS voice memo), FFmpeg will automatically resample to the nearest supported AC3 rate. The resulting .ac3 file is an elementary stream and lacks the container-level metadata that formats like MP4 or MKV provide, so it is best suited for direct muxing into DVD, Blu-ray, or broadcast container workflows rather than standalone distribution.