Convert CAF to AIFC — Free Online Tool
Convert CAF (Core Audio Format) files to AIFC with lossless PCM audio using FFmpeg in your browser. This tool transcodes the audio stream from CAF's little-endian PCM to AIFC's big-endian PCM_S16BE format, making your Apple audio files compatible with professional DAWs and legacy systems that expect AIFF-family containers.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
CAF stores audio in little-endian byte order (PCM_S16LE by default), while AIFC — an extension of Apple's classic AIFF format — requires big-endian byte order (PCM_S16BE). This conversion re-encodes the raw audio samples, flipping the byte order from little-endian to big-endian while preserving the full 16-bit depth and sample rate. The AIFC container wraps the resulting PCM_S16BE stream in a format that is readable by professional audio workstations, older Mac software, and any tool that understands the AIFF/AIFC specification. Because both formats carry uncompressed PCM audio, there is no perceptual quality loss — only a byte-order transformation occurs.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion entirely inside your browser via WebAssembly (FFmpeg.wasm). |
-i input.caf
|
Specifies the input file as a CAF (Core Audio Format) container. FFmpeg reads the CAF header to detect the audio codec, sample rate, bit depth, and channel layout before processing. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed PCM in big-endian byte order — the native PCM format of the AIFC specification. This reverses the byte order from CAF's little-endian PCM_S16LE, making the data readable by any AIFF/AIFC-compliant application. |
-b:a 128k
|
Passes a 128 kbps bitrate hint to the audio encoder. For PCM_S16BE this parameter has no practical effect since uncompressed PCM bitrate is fixed by sample rate and bit depth, but it is included for consistency with the tool's interface and is safely ignored by the encoder. |
output.aifc
|
Defines the output filename and tells FFmpeg to wrap the encoded PCM_S16BE audio stream in an AIFC container. The '.aifc' extension is the standard designation for AIFF-C files and is recognised natively by macOS Core Audio and most professional DAWs. |
Common Use Cases
- Importing Apple-recorded CAF audio into a professional DAW like Pro Tools or Logic Pro 9 that expects AIFF-family files rather than CAF containers
- Preparing audio for broadcast or archival workflows that mandate big-endian PCM storage as specified by legacy AIFF/AIFC interchange standards
- Moving voice memos or Core Audio recordings from macOS/iOS into audio editing software on older Mac systems that do not support the CAF container
- Converting CAF test recordings from Apple's AVAudioRecorder into AIFC for use with audio analysis tools that parse AIFF chunk structures
- Migrating a library of CAF-based sound effects or samples to AIFC so they can be loaded by samplers and instruments that predate CAF support
- Producing AIFC deliverables from CAF source files for post-production pipelines that archive finished audio in AIFF-compatible formats
Frequently Asked Questions
No — this specific conversion is lossless in terms of audio fidelity. Both the input (PCM_S16LE in CAF) and the output (PCM_S16BE in AIFC) are uncompressed 16-bit PCM formats; only the byte order of the raw samples is reversed. No compression artifacts, no frequency-range clipping, and no dynamic-range reduction occur during this process.
AIFC inherits the big-endian convention from the original AIFF specification, which was designed in the late 1980s for Motorola 68000-based Macs that used big-endian architecture. CAF, introduced by Apple in 2005, adopted little-endian byte order to align with the Intel x86 processors Apple had transitioned to. The two formats are therefore byte-order mirrors of each other for raw PCM data.
Sample rate and channel count are preserved because FFmpeg reads them from the CAF container and writes them into the AIFC header. However, CAF supports rich metadata chunks (such as iTunes-style tags) that have no direct equivalent in the AIFC specification, so text metadata like track titles or artist names may be dropped during the conversion. Verify metadata in your target application after converting.
Yes — if your CAF file contains AAC, FLAC, Opus, or Vorbis audio, FFmpeg will decode it to PCM first and then write PCM_S16BE into the AIFC container. The command remains the same. Keep in mind that transcoding from a lossy format like AAC into PCM does not recover lost audio information; it simply stores the already-decoded signal losslessly in AIFC.
Replace '-c:a pcm_s16be' with '-c:a pcm_s24be' for 24-bit big-endian PCM output. AIFC also supports pcm_s32be and pcm_f32be for 32-bit integer and 32-bit float output respectively. If your source CAF file was recorded at 24-bit depth (pcm_s24le), using pcm_s24be in the output command preserves the original bit depth without any truncation: 'ffmpeg -i input.caf -c:a pcm_s24be output.aifc'.
On macOS or Linux you can run a shell loop: 'for f in *.caf; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.caf}.aifc"; done'. On Windows Command Prompt use: 'for %f in (*.caf) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc"'. This applies the same PCM_S16BE encoding to every CAF file in the current directory and outputs a matching AIFC file alongside each source.
Technical Notes
CAF and AIFC are both Apple-ecosystem formats but occupy different eras of the platform's history. CAF was engineered to remove the 4 GB file-size ceiling of AIFF and to natively support a wide codec roster including AAC, FLAC, Opus, and Vorbis — which is why it is the default recording container on iOS and macOS. AIFC, by contrast, is a more constrained format: it stores only PCM variants (s16be, s24be, s32be, f32be, f64be) and the telephony codecs A-law and µ-law, all in big-endian byte order. This means that any lossy audio inside a CAF file (AAC being the most common) must be fully decoded to PCM before it can be stored in AIFC, potentially increasing file size substantially. For a 16-bit, 44.1 kHz stereo source recorded as PCM_S16LE in CAF, the resulting AIFC will be virtually identical in size. The '-b:a 128k' flag in the default command is effectively unused for PCM codecs — bitrate is determined by sample rate, bit depth, and channel count — so removing it has no effect on PCM output. AIFC files use the '.aifc' or '.aif' extension and are read natively by macOS QuickTime, Core Audio, and most professional DAWs; however, some Linux and Windows audio tools may require a plugin or updated decoder to handle AIFC as distinct from plain AIFF.