Convert WAV to AIFC — Free Online Tool
Convert WAV audio files to AIFC format, transcoding from little-endian PCM (the Windows standard) to big-endian PCM as used in Apple's AIFC container. This is the go-to conversion for moving uncompressed audio from Windows production environments into Apple-native professional workflows while preserving full lossless fidelity.
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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
WAV files store PCM audio in little-endian byte order, a convention inherited from Intel x86 architecture and the Windows ecosystem. AIFC (an extension of Apple's AIFF format) stores PCM audio in big-endian byte order, reflecting its Motorola 68000 heritage. During this conversion, FFmpeg reads the raw PCM samples from the WAV container and rewrites them with reversed byte order into the AIFC container using the pcm_s16be codec — 'be' standing for big-endian. The actual audio sample values are mathematically identical; only the byte ordering and container wrapper change. No lossy compression is applied, so the conversion is bit-for-bit equivalent in terms of audio fidelity. The AIFC format also supports optional compressed audio codecs like A-law and µ-law, but this conversion targets uncompressed 16-bit PCM for maximum compatibility with professional Apple audio tools.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm), so no audio data leaves your device. |
-i input.wav
|
Specifies the input WAV file. FFmpeg detects the WAV container and identifies the audio stream's codec (typically pcm_s16le — 16-bit little-endian PCM) and parameters such as sample rate and channel count. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM, which is the native uncompressed format for AIFC. This performs a lossless byte-order swap from WAV's little-endian samples to AIFC's big-endian samples — no audio data is lost or compressed. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps, but this parameter has no practical effect when the codec is uncompressed PCM (pcm_s16be). The actual bitrate of the output is fixed by sample rate and bit depth — for a standard 44.1 kHz stereo file it will be approximately 1,411 kbps. |
output.aifc
|
Defines the output filename and triggers FFmpeg to wrap the re-encoded big-endian PCM audio in the AIFC container format, which uses Apple's chunk-based file structure derived from the original AIFF specification. |
Common Use Cases
- Delivering audio assets to a post-production studio running Pro Tools on macOS, which historically expects AIFF/AIFC files over WAV for certain session types
- Archiving field recordings originally captured as WAV on a Windows laptop into AIFC for ingestion into a Logic Pro X project on a Mac
- Preparing sound effects or music beds in AIFC format for a video editor working in Final Cut Pro, which has native AIFC support
- Converting WAV broadcast recordings to AIFC for submission to a broadcaster or distributor whose delivery specification requires Apple-native audio containers
- Migrating a Windows-based audio library into an AIFC-centric macOS workflow without any loss of audio quality or need for re-recording
- Testing cross-platform audio pipeline compatibility by converting a reference WAV file to AIFC and verifying sample-accurate playback in both environments
Frequently Asked Questions
No — this specific conversion is entirely lossless. Both the input WAV (pcm_s16le) and the output AIFC (pcm_s16be) store uncompressed 16-bit PCM audio; the only difference is byte order. FFmpeg reverses the byte ordering of each sample, but the numerical audio values are identical. You will not hear any difference in a null test or ABX comparison.
Both codecs store 16-bit signed integer PCM audio samples, but they differ in how the two bytes of each sample are ordered in the file. pcm_s16le (little-endian) stores the least significant byte first, as is standard on x86/Windows systems. pcm_s16be (big-endian) stores the most significant byte first, as used by older Motorola processors and preserved in the AIFF/AIFC family. To any audio playback system that correctly decodes the container, the sound is identical.
Sample rate, bit depth, and channel count are always preserved because they are core stream properties that FFmpeg carries through automatically. However, embedded metadata tags (ID3-style tags sometimes added to WAV files) may not transfer cleanly into AIFC, since AIFC uses its own chunk-based metadata structure. For critical metadata preservation, review the output file in an audio editor like Audacity or Adobe Audition after conversion.
The '-b:a 128k' flag is included as a default parameter in the command template, but when the audio codec is set to pcm_s16be (uncompressed PCM), FFmpeg ignores the bitrate setting — PCM audio has a fixed bitrate determined entirely by sample rate and bit depth, not a configurable compression target. For a 44.1 kHz stereo 16-bit file, the actual bitrate will always be approximately 1,411 kbps regardless of the '-b:a' value. You can safely omit that flag when running the command locally.
Replace 'pcm_s16be' in the '-c:a' flag with another supported AIFC PCM codec. For 24-bit output use 'pcm_s24be', for 32-bit integer use 'pcm_s32be', for 32-bit float use 'pcm_f32be', and for 64-bit float use 'pcm_f64be'. For example: 'ffmpeg -i input.wav -c:a pcm_s24be output.aifc'. If your source WAV is already 24-bit (pcm_s24le), converting to pcm_s24be preserves the full dynamic range of the original recording.
Yes. On macOS or Linux, you can loop over all WAV files in a directory with: 'for f in *.wav; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.wav}.aifc"; done'. On Windows Command Prompt, use: 'for %f in (*.wav) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc"'. The browser-based tool processes one file at a time, so the FFmpeg command is particularly valuable for bulk conversions of large libraries.
Technical Notes
The WAV-to-AIFC conversion is fundamentally a container remux with a byte-order swap rather than a lossy transcode. WAV's default codec (pcm_s16le) and AIFC's default codec (pcm_s16be) are mirror images of each other: same sample width, same signedness, opposite endianness. One subtle technical consideration is file size — AIFC files converted this way will be nearly identical in size to the source WAV, with only minor differences from container overhead (AIFC chunk headers versus WAV's RIFF header structure). AIFC also supports µ-law and A-law compressed codecs, which are useful for telephony applications, but this tool targets the lossless PCM path. Note that AIFC is technically distinct from plain AIFF: the 'C' suffix indicates the container supports compression markers, even when storing uncompressed PCM. Most professional macOS audio applications (Logic Pro, Pro Tools, Audacity, Final Cut Pro) handle AIFC natively. One known limitation is that WAV files containing RF64 chunks (used for files over 4 GB) may not parse correctly in all AIFC-reading applications after conversion, so splitting very large WAV files before conversion is advisable.