Convert AIF to WAV — Free Online Tool
Convert AIF files to WAV by transcoding Apple's big-endian PCM audio (pcm_s16be) into the little-endian PCM format (pcm_s16le) that Windows, Linux, and virtually every DAW and audio application expect. Both formats are lossless and uncompressed, so this conversion preserves full audio fidelity while making your files universally compatible.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AIF 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
AIF and WAV are both uncompressed PCM audio containers, but they differ in byte order: AIF uses big-endian byte ordering (a legacy of Motorola-based Macs), while WAV uses little-endian byte ordering (the x86/Windows standard). FFmpeg reads the AIF container, decodes the pcm_s16be (16-bit big-endian PCM) audio stream, and rewrites the raw PCM samples in little-endian order (pcm_s16le) inside a WAV RIFF container. Because no compression or lossy encoding is applied, the audio waveform data itself is identical — only the byte order and container wrapper change. The resulting WAV file will be approximately the same size as the source AIF file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In this browser-based tool, FFmpeg runs as a WebAssembly module (FFmpeg.wasm) entirely within your browser — no files leave your device. When running locally on your desktop, this calls your system-installed FFmpeg binary. |
-i input.aif
|
Specifies the input file — your source AIF audio file. FFmpeg reads the AIFF/IFF container, identifies the audio stream (typically pcm_s16be, big-endian 16-bit PCM), and parses the COMM chunk for sample rate and channel information. |
-c:a pcm_s16le
|
Sets the output audio codec to pcm_s16le — signed 16-bit little-endian PCM, which is the standard uncompressed audio encoding for WAV files on x86/Windows systems. This is a lossless byte-order conversion from AIF's big-endian PCM (pcm_s16be); no compression or quality reduction is applied. |
output.wav
|
Defines the output filename and tells FFmpeg to wrap the pcm_s16le audio stream in a WAV RIFF container. The .wav extension triggers FFmpeg's WAV muxer, which writes the standard 'RIFF'/'fmt '/'data' chunk structure expected by Windows applications, DAWs, and broadcast audio tools. |
Common Use Cases
- Sending audio stems or samples to a collaborator using a Windows-based DAW like FL Studio or older versions of Audacity that do not reliably parse AIF files.
- Loading Apple Logic Pro or GarageBand exports into video editing software like DaVinci Resolve or Adobe Premiere that prefers WAV for timeline audio tracks.
- Preparing sound effects or field recordings originally captured on a Mac for use in a Windows game engine (Unity, Unreal) or middleware like FMOD or Wwise.
- Archiving or delivering broadcast audio that must conform to WAV specifications required by radio stations, post-production facilities, or podcast distribution platforms.
- Converting a library of AIF samples purchased from a Mac-centric sound library so they can be dragged into an Ableton Live, Bitwig, or Reaper session on any platform without format warnings.
- Running automated audio pipelines or scripts on Linux servers where AIF support is inconsistent but WAV (pcm_s16le) is natively handled by nearly every audio tool.
Frequently Asked Questions
No. Both AIF (using pcm_s16be) and WAV (using pcm_s16le) store raw, uncompressed PCM audio. The only change is the byte order of the samples — big-endian to little-endian — which is a lossless mathematical transformation. Every sample value is preserved exactly, so a null test between the source and converted file would show no audible or measurable difference.
AIF was developed by Apple in the 1980s for Motorola 68000-based Macs, which used big-endian architecture, so audio data is stored with the most significant byte first. WAV was developed by Microsoft for x86 PCs, which use little-endian architecture, storing the least significant byte first. Modern software handles this transparently during playback, but many applications that read raw audio streams or use strict format parsing will reject or misinterpret files with the wrong byte order, which is why this conversion is necessary.
AIF stores metadata in its MARK, INST, and ID3-style chunks, while WAV uses LIST/INFO and SMPL chunks for equivalent data. FFmpeg will attempt to map common metadata fields (title, artist, etc.) from the AIF container to WAV INFO chunks, but format-specific data like AIF loop markers or instrument definitions are unlikely to be preserved in the output. If loop points or sampler metadata are critical, verify them in your target application after conversion.
This tool applies the FFmpeg command with -c:a pcm_s16le, which converts the audio to 16-bit WAV regardless of the source bit depth. If your AIF file contains 24-bit (pcm_s24be) or 32-bit audio and you need to preserve that bit depth, you should run FFmpeg locally and change the codec flag: use -c:a pcm_s24le for 24-bit output or -c:a pcm_s32le for 32-bit output. The FFmpeg command displayed on this page makes it easy to adapt for higher bit depths.
You can adapt the displayed command into a shell loop. On macOS or Linux, run: for f in *.aif; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.aif}.wav"; done. On Windows Command Prompt, use: for %f in (*.aif) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This applies the same lossless big-endian to little-endian conversion to every AIF file in the current directory.
The audio data size will be virtually identical because both formats store uncompressed PCM at the same bit depth and sample rate. Any minor size difference comes from the container header and metadata chunks — the AIFF header versus the WAV RIFF header — which are only a few dozen bytes. You should not expect any meaningful file size increase or decrease from this conversion.
Technical Notes
AIF and WAV occupy the same conceptual space — both are uncompressed PCM containers — but their internal structure differs enough to cause real-world compatibility issues. AIF wraps audio in an IFF (Interchange File Format) chunk structure with a COMM chunk defining sample rate, bit depth, and channel count, while WAV uses Microsoft's RIFF container with a 'fmt ' chunk. The key codec difference is endianness: FFmpeg labels AIF's default codec as pcm_s16be and WAV's as pcm_s16le. The conversion using -c:a pcm_s16le is lossless in the strict sense — no frequency content, dynamic range, or sample values are altered. One practical limitation is bit depth: this command standardizes output to 16-bit, which is appropriate for CD-quality audio but will truncate the dynamic range of 24-bit or 32-bit AIF sources. AIF supports higher bit-depth PCM variants (pcm_s24be, pcm_s32be, pcm_f32be, pcm_f64be), and WAV has corresponding little-endian equivalents (pcm_s24le, pcm_s32le, pcm_f32le) that can be targeted by modifying the -c:a flag. Metadata portability is partial: FFmpeg maps standard tags but cannot guarantee preservation of AIF-specific MARK or INST chunks in the WAV output.