Convert WAV to AIF — Free Online Tool
Convert WAV audio files to AIF format by repackaging PCM audio data from Microsoft's little-endian WAV container into Apple's big-endian AIFF container — with no quality loss. This tool runs entirely in your browser using FFmpeg.wasm, so your audio files never leave your device.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your WAV 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
WAV and AIF are both uncompressed PCM audio formats, so this conversion involves no re-encoding of the audio data itself — just a container swap with a byte-order change. WAV stores PCM samples in little-endian byte order (least significant byte first), while AIF uses big-endian byte order (most significant byte first). FFmpeg reads the raw PCM samples from the WAV container and writes them into the AIF container using the pcm_s16be codec, flipping the byte order in the process. The result is bit-for-bit identical audio content in a format that Apple software and macOS workflows expect natively. Because no lossy compression is applied at any stage, there is zero generational quality loss.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop command line. |
-i input.wav
|
Specifies the input file — a WAV container that typically holds PCM audio in little-endian byte order, as produced by Windows recording software, DAWs, or audio interfaces on x86 systems. |
-c:a pcm_s16be
|
Sets the output audio codec to signed 16-bit big-endian PCM, which is the native uncompressed audio encoding used inside AIF files. This swaps the byte order from WAV's little-endian layout to AIFF's big-endian layout without changing the actual audio samples or introducing any lossy compression. |
output.aif
|
Defines the output filename with the .aif extension, which tells FFmpeg to wrap the pcm_s16be audio stream in an AIFF container — Apple's Audio Interchange File Format — compatible with macOS, Logic Pro, GarageBand, and other Apple audio tools. |
Common Use Cases
- Delivering audio assets to a Mac-based audio engineer or music producer whose DAW (such as Logic Pro) defaults to or requires AIFF files
- Preparing sound effects or music stems recorded on a Windows workstation (as WAV) for use in an Apple-centric post-production pipeline
- Archiving recordings in AIFF format for compatibility with legacy Mac software or hardware samplers like the Akai S-series that reads AIF natively
- Converting Windows-produced voice-over or podcast raw recordings from WAV to AIF before importing into GarageBand or Final Cut Pro on a Mac
- Supplying broadcast-ready audio to a media outlet whose submission specifications require AIFF rather than WAV format
- Migrating a library of WAV samples to AIF so they display and preview correctly in macOS Finder without relying on third-party codec support
Frequently Asked Questions
No — this conversion is completely lossless. Both WAV and AIF store audio as uncompressed PCM, and the conversion only changes the container format and byte order (little-endian to big-endian). The actual audio samples are mathematically identical before and after conversion. You could convert back to WAV and get bit-perfect audio.
The byte-order difference reflects the computing architectures each format was designed for. WAV was developed by Microsoft and IBM for x86 PCs, which use little-endian byte order. AIF was developed by Apple for Motorola 68000-based Macs, which use big-endian byte order. Modern Macs (even on Apple Silicon) can handle both, but AIF remains the native Apple format and is the default expectation in macOS audio tools.
Yes. FFmpeg carries over the sample rate, channel count, and bit depth from the source WAV file into the AIF container. If your WAV is stereo 44.1 kHz 16-bit, your AIF output will also be stereo 44.1 kHz 16-bit. However, embedded text metadata such as artist or title tags stored in WAV's LIST/INFO or ID3 chunks may not transfer, as AIF uses a different metadata chunk structure (ANNO and NAME chunks).
The FFmpeg command shown on this page uses pcm_s16be, which will convert a 24-bit WAV to a 16-bit AIF. If you need to preserve 24-bit depth, you should modify the command to use pcm_s24be instead: ffmpeg -i input.wav -c:a pcm_s24be output.aif. You can copy this modified command and run it locally in your terminal — this is one reason the FFmpeg command is displayed on the page.
On macOS or Linux, you can use a shell loop: for f in *.wav; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.wav}.aif"; done. On Windows Command Prompt, use: for %f in (*.wav) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". This applies the same pcm_s16be codec used by this browser tool to every WAV file in the current directory.
Yes. AIFF with pcm_s16be is one of the most natively supported formats across all major Mac DAWs. Logic Pro and GarageBand have treated AIFF as a first-class citizen since their earliest versions. Pro Tools also reads AIFF without issue, though it often prefers BWF-extended WAV for session work. For sample import, loop libraries, or audio asset delivery on macOS, AIF is an excellent choice.
Technical Notes
The core technical distinction between WAV and AIF in this conversion is endianness: WAV's pcm_s16le (signed 16-bit little-endian) becomes AIF's pcm_s16be (signed 16-bit big-endian). FFmpeg handles this byte-swap transparently. File sizes will be nearly identical between the source WAV and output AIF, as both are uncompressed — the only size difference comes from differing container header overhead, which is negligible. One notable limitation is metadata handling: WAV files sometimes carry broadcast WAV (BWF) metadata in a 'bext' chunk, which has no direct equivalent in standard AIFF and will be dropped during conversion. If your WAV source contains audio in a compressed format like ADPCM or FLAC-in-WAV, FFmpeg will need to decode those samples first before re-encoding them as pcm_s16be for the AIF output, which is still lossless relative to the original uncompressed signal but involves an intermediate decode step. AIF does not support chapters, subtitles, or multiple audio tracks, and neither does WAV, so no features are lost in that regard.