Convert AIFC to WAV — Free Online Tool
Convert AIFC files to WAV by transcoding from Apple's big-endian PCM or compressed audio format to Microsoft's universally compatible little-endian PCM WAV format. This conversion is essential for moving professional Apple audio into workflows that require standard uncompressed WAV files without codec dependencies.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AIFC 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
AIFC stores audio in big-endian byte order (most significant byte first), which is native to classic Motorola/PowerPC architectures used in older Apple hardware. WAV uses little-endian byte order (least significant byte first), native to x86/Intel systems. During this conversion, FFmpeg reads the AIFC container — which may contain uncompressed PCM variants like pcm_s16be, pcm_s24be, or pcm_s32be, or compressed codecs like pcm_alaw and pcm_mulaw — and re-encodes the audio stream as pcm_s16le, the standard 16-bit signed little-endian PCM used by WAV. This is a full audio re-encode, not a remux, because the byte order and container structure are incompatible. The output is an uncompressed WAV file with broad compatibility across all modern operating systems, DAWs, and audio tools.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — in this browser-based tool, it runs as a WebAssembly (FFmpeg.wasm) instance entirely within your browser, with no file upload to any server. |
-i input.aifc
|
Specifies the input file in AIFC format. FFmpeg reads the AIFC container, identifies the audio codec (which may be big-endian PCM or a compressed variant like pcm_alaw), and prepares to decode it for re-encoding. |
-c:a pcm_s16le
|
Sets the output audio codec to 16-bit signed little-endian PCM, which is the standard uncompressed audio codec for WAV files on x86/Intel systems. This byte-swaps the data from AIFC's big-endian format and produces a WAV file that is universally readable across all modern platforms. |
output.wav
|
Specifies the output filename with the .wav extension. FFmpeg uses this extension to select the RIFF WAV container format, which wraps the pcm_s16le audio stream in a header that includes sample rate, channel count, and bit depth metadata readable by any WAV-compatible application. |
Common Use Cases
- Opening legacy Mac or Pro Tools audio sessions on Windows-based DAWs that cannot parse AIFC's big-endian byte order or compressed variants
- Preparing AIFC audio recorded on classic Apple hardware or older Logic Pro versions for use in Windows-native software like Adobe Audition or Sound Forge
- Converting AIFC files that use compressed codecs like G.711 A-law or μ-law (common in telephony) into plain uncompressed WAV for archival or editing purposes
- Delivering audio assets to video editors or broadcast facilities whose ingest systems require standard little-endian PCM WAV and do not support AIFC containers
- Batch-converting a library of AIFC samples or field recordings from an older Apple-based recording rig into WAV for use in modern sample libraries and plugin instruments
- Debugging or inspecting AIFC audio content by converting to WAV so the file can be opened in any basic audio player or waveform viewer without needing AIFC-aware software
Frequently Asked Questions
If your AIFC file uses an uncompressed PCM codec such as pcm_s16be, the conversion to pcm_s16le WAV is mathematically lossless — only the byte order changes and no audio data is discarded. However, if your AIFC uses a compressed codec like pcm_alaw or pcm_mulaw, those codecs are inherently lossy and the WAV output will reflect the quality of the already-compressed source. Converting a 24-bit or 32-bit AIFC down to the default 16-bit WAV output will also reduce bit depth, so you may want to adjust the FFmpeg command if preserving the original bit depth matters.
AIFC was developed by Apple in the era of Motorola 68000 and PowerPC processors, which natively processed data in big-endian order (most significant byte first). WAV was developed by Microsoft and IBM for x86-based PCs, which use little-endian order (least significant byte first). These are simply different conventions baked into each format's specification. FFmpeg handles the byte-swapping transparently during conversion, so the audio output is bit-perfect relative to the source, just stored in the byte order WAV expects.
The default FFmpeg command outputs 16-bit WAV (pcm_s16le). To preserve 24-bit depth from a pcm_s24be AIFC source, change the audio codec flag to '-c:a pcm_s24le'. For 32-bit float AIFC (pcm_f32be), use '-c:a pcm_f32le'. For example: 'ffmpeg -i input.aifc -c:a pcm_s24le output.wav'. This ensures no bit-depth reduction occurs and the WAV file retains the full dynamic range of the original recording.
AIFC supports metadata through its MARK and NAME chunks, but WAV uses a different metadata structure (INFO chunks or BEXT chunks for broadcast WAV). FFmpeg will attempt to map common metadata fields such as title and artist, but format-specific markers, loop points, or instrument chunk data embedded in the AIFC will not transfer to WAV. If metadata preservation is critical, verify the output file in your target application after conversion.
Yes. On Linux or macOS, you can run a shell loop: 'for f in *.aifc; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.aifc}.wav"; done'. On Windows Command Prompt, use: 'for %f in (*.aifc) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav"'. This applies the same pcm_s16le encoding to every AIFC file in the current directory, outputting a corresponding WAV file for each.
If your AIFC used a compressed codec like pcm_alaw or pcm_mulaw, the WAV output will be significantly larger because those codecs compress audio data and the WAV output is fully uncompressed pcm_s16le. Conversely, if the AIFC was already uncompressed 16-bit PCM (pcm_s16be), the file sizes will be nearly identical since only the container header and byte order change. A 24-bit or 32-bit AIFC converted to 16-bit WAV will be smaller due to the reduction in bit depth.
Technical Notes
AIFC (Audio Interchange File Format Compressed) is a superset of Apple's AIFF format, adding support for compressed audio codecs beyond uncompressed PCM. The format stores multi-byte samples in big-endian order, which is directly incompatible with WAV's little-endian requirement — making a straightforward remux impossible. FFmpeg must decode and re-encode the audio stream even when both source and destination are technically PCM, because the byte order must be reversed sample by sample. The default conversion targets pcm_s16le at the source file's original sample rate and channel count, preserving those parameters while changing bit depth to 16-bit if the source was higher. WAV has a theoretical file size limit of 4GB due to its 32-bit RIFF chunk size field, though most modern applications support the RF64 extension for larger files. AIFC's less common codecs — such as MACE 3:1 and MACE 6:1 compression used in very old Mac software — may not be supported by all FFmpeg builds. The resulting WAV files are broadly compatible with virtually all professional and consumer audio software, DAWs, broadcast ingest systems, and operating system audio APIs without any additional codec installation.