Convert CAF to AIFF — Free Online Tool
Convert CAF audio files to AIFF format by re-encoding the audio stream to uncompressed PCM big-endian — perfect for moving Apple Core Audio files into a universally readable lossless format compatible with professional audio workstations and legacy Mac software.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your CAF 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
CAF files can contain audio encoded in a variety of codecs including AAC, FLAC, Opus, Vorbis, or various PCM formats. AIFF, by contrast, is a strictly uncompressed format that only supports big-endian PCM audio. During this conversion, FFmpeg decodes whatever audio codec is present in the CAF container and re-encodes it to 16-bit signed big-endian PCM (pcm_s16be), which is the standard codec for AIFF. If your CAF file already contains 16-bit PCM audio, the audio is decoded and re-encoded into the new byte order and container structure. The result is a fully uncompressed AIFF file with broad compatibility across macOS applications, DAWs, and professional audio tools. Note that if the source CAF contained lossy audio (such as AAC or Opus), the AIFF output will be lossless in format but cannot recover quality lost during the original lossy encoding.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no audio data leaves your device. |
-i input.caf
|
Specifies the input file, which is a CAF (Core Audio Format) container. FFmpeg will detect the audio codec inside the CAF file automatically — whether that is pcm_s16le, AAC, FLAC, or another supported codec — and decode it accordingly. |
-c:a pcm_s16be
|
Sets the audio codec for the output to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding used in AIFF files. This differs from CAF's default little-endian PCM, so FFmpeg reverses the byte order of each audio sample during encoding — a lossless transformation when the source is also 16-bit PCM. |
output.aiff
|
Specifies the output filename with the .aiff extension, which tells FFmpeg to write the result in the Audio Interchange File Format container. The AIFF container wraps the pcm_s16be audio stream in the format expected by macOS applications, DAWs, and professional audio tools. |
Common Use Cases
- Opening an iPhone or iOS voice memo or recorded audio (saved as CAF) in a professional DAW like Logic Pro or Pro Tools that expects AIFF input
- Archiving CAF files containing uncompressed PCM audio into AIFF for long-term storage in a widely supported, non-proprietary lossless format
- Preparing audio assets recorded on macOS in CAF format for use in video editing software such as Final Cut Pro or Adobe Premiere that handles AIFF more reliably
- Converting CAF audio samples or loops for use in sample libraries and virtual instruments that only accept AIFF or WAV files
- Sharing lossless audio files with collaborators or studios on systems where CAF is not natively supported but AIFF is a standard interchange format
Frequently Asked Questions
No — if your CAF file contains AAC-encoded audio, the quality lost during AAC encoding cannot be recovered by converting to AIFF. The output AIFF will be an uncompressed file, but it will contain audio decoded from the AAC stream, preserving the artifacts and frequency limitations introduced when the audio was originally compressed. For a true lossless result, the conversion needs to start from a CAF file containing lossless audio such as pcm_s16le, FLAC, or another lossless source.
AIFF was designed by Apple in 1988 for Motorola 68k processors, which use big-endian byte ordering, and the format has retained that convention ever since. CAF, introduced much later in 2005, defaults to little-endian PCM (pcm_s16le) to align with modern Intel and ARM processor architectures. This means even when both the CAF source and AIFF output store audio as 16-bit PCM, FFmpeg must reverse the byte order of each sample during conversion — pcm_s16le to pcm_s16be — which is a lightweight transformation with no quality impact.
It depends on what codec the CAF file was using. If the CAF contained compressed audio such as AAC, Opus, or Vorbis, the resulting AIFF will be significantly larger because AIFF is always uncompressed. If the CAF already contained uncompressed 16-bit PCM audio, the file size will be approximately the same, with minor differences due to container overhead and metadata structures. CAF files with FLAC audio will also expand in size when converted to uncompressed AIFF.
CAF supports metadata chunks, but AIFF has a limited and older metadata specification that does not map cleanly to all CAF metadata fields. FFmpeg will attempt to carry over compatible metadata, but fields specific to CAF or fields using formats AIFF cannot accommodate may be dropped. If preserving detailed metadata is important, verify the output file's tags after conversion using a tool like MediaInfo or an audio tag editor.
Replace pcm_s16be in the command with a different PCM variant to change the bit depth. Use pcm_s24be for 24-bit output, pcm_s32be for 32-bit integer, pcm_f32be for 32-bit float, or pcm_f64be for 64-bit float. For example: ffmpeg -i input.caf -c:a pcm_s24be output.aiff. If your source CAF contains 24-bit audio, outputting as pcm_s24be preserves the full resolution rather than downsampling to 16-bit.
Yes — on macOS or Linux you can use a shell loop to process multiple files at once. For example: for f in *.caf; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.caf}.aiff"; done. This iterates over every CAF file in the current directory and produces a matching AIFF file. On Windows, a similar loop can be written in PowerShell or a batch script. The browser-based tool processes one file at a time, but the FFmpeg command displayed on this page is ideal for bulk conversion tasks on your local machine.
Technical Notes
CAF is a highly flexible container that can encapsulate many different audio codecs, making it important to know your source file's actual codec before conversion. AIFF, by contrast, is strictly a PCM container — all five codecs it supports (pcm_s16be, pcm_s24be, pcm_s32be, pcm_f32be, pcm_f64be) are uncompressed big-endian variants. This tool defaults to pcm_s16be, which is the AIFF standard and broadly supported, but users with high-resolution CAF sources (24-bit or 32-bit PCM, or FLAC) should consider modifying the command to use pcm_s24be or pcm_s32be to avoid downsampling the bit depth. CAF supports channel layouts and sample rates beyond what many formats handle, including multi-channel configurations, but AIFF's multi-channel support in practice depends on the receiving application. The sample rate from the source CAF is preserved by FFmpeg unless explicitly changed with the -ar flag. AIFF does not support subtitles, chapters, or embedded album art in the way modern containers do, so any such metadata present in a CAF file will be discarded during conversion. For professional audio archiving workflows, AIFF at 24-bit or higher is generally preferred over the default 16-bit output this command produces.