Convert WAV to AIFF — Free Online Tool
Convert WAV files to AIFF format using a lossless PCM transcode — switching from little-endian (pcm_s16le) to big-endian (pcm_s16be) byte order while preserving every bit of your original audio. Ideal for moving uncompressed audio between Windows and macOS professional environments without any quality loss.
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FFmpeg Command
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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 and AIFF are both uncompressed PCM audio containers, but they differ in byte order and platform origin. WAV stores PCM audio in little-endian format (developed by Microsoft), while AIFF stores it in big-endian format (developed by Apple). During this conversion, FFmpeg reads the raw PCM samples from the WAV container and rewrites them into the AIFF container using the pcm_s16be codec — reversing the byte order of each 16-bit sample. No audio data is discarded, no lossy compression is applied, and no resampling occurs. The result is a bit-perfect representation of the original audio in a format natively understood by macOS applications like Logic Pro, GarageBand, and Final Cut Pro.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles the WAV parsing, PCM byte-order conversion, and AIFF container writing in this command. |
-i input.wav
|
Specifies the input WAV file. FFmpeg reads the RIFF container, identifies the PCM audio stream (typically pcm_s16le for standard WAV), and prepares it for decoding into raw audio samples. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, which is the native PCM encoding used inside AIFF containers. This reverses the byte order of each audio sample from WAV's little-endian layout, producing identical audio data in the format AIFF expects. |
output.aiff
|
Defines the output filename and triggers AIFF container formatting. FFmpeg uses the .aiff extension to automatically wrap the pcm_s16be audio stream in Apple's AIFF chunk structure, producing a file natively compatible with macOS audio applications. |
Common Use Cases
- Delivering audio assets to a macOS-based music producer or post-production studio that requires AIFF files for Logic Pro sessions
- Converting WAV recordings from a Windows DAW like Reaper or Audition into AIFF so they can be imported into GarageBand or Final Cut Pro without format warnings
- Preparing broadcast-ready audio stems in AIFF format, which is commonly required by broadcast facilities and Apple platforms over WAV
- Archiving WAV field recordings from a portable recorder into AIFF for long-term storage in an Apple-centric media library
- Migrating a sample library from a Windows machine to a Mac without re-encoding or degrading any audio samples
- Converting WAV sound effects files to AIFF for use in Xcode projects or iOS/macOS app development, where AIFF is a natively supported audio format
Frequently Asked Questions
No — this is a fully lossless conversion. Both WAV (using pcm_s16le) and AIFF (using pcm_s16be) store raw, uncompressed PCM audio. The only change is the byte order of the PCM samples, which is a mathematical reversal with no rounding errors or discarded data. Your waveform is identical before and after conversion.
Although WAV and AIFF both contain PCM audio, they use opposite byte orders — WAV uses little-endian (pcm_s16le) and AIFF uses big-endian (pcm_s16be). A raw stream copy (-c:a copy) would paste little-endian bytes into a big-endian container, producing corrupted audio. FFmpeg must explicitly decode and re-encode the samples with pcm_s16be to correctly reverse the byte order for AIFF.
The audio data size will be nearly identical, since both formats store uncompressed 16-bit PCM at the same bit depth and sample rate. AIFF and WAV have slightly different container overhead and metadata structures, so you may see a marginal difference of a few kilobytes, but there is no meaningful change in file size.
Sample rate and channel count are always preserved because they are fundamental PCM stream properties. However, text metadata tags (such as artist or title embedded in WAV's INFO or ID3 chunks) may not transfer cleanly to AIFF, since the two containers use different metadata schemes — WAV uses RIFF INFO chunks and AIFF uses its own ANNO and NAME chunks. If precise metadata preservation is critical, verify the output file's tags after conversion.
Change the codec in the FFmpeg command to match your source bit depth. For a 24-bit WAV (pcm_s24le), use '-c:a pcm_s24be'; for 32-bit integer WAV (pcm_s32le), use '-c:a pcm_s32be'; for 32-bit float WAV (pcm_f32le), use '-c:a pcm_f32be'. The default command uses pcm_s16be, which is correct for standard 16-bit WAV files.
Yes. On Linux or macOS, you can run a shell loop: 'for f in *.wav; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.wav}.aiff"; done'. On Windows Command Prompt, use: 'for %f in (*.wav) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff"'. This applies the same lossless byte-order conversion to every WAV file in the current directory.
Technical Notes
WAV and AIFF are architecturally similar — both are chunk-based container formats wrapping raw PCM audio — but their divergent platform heritage means they handle byte order differently. WAV's PCM data is little-endian (least significant byte first), a convention inherited from Intel x86 architecture, while AIFF's PCM data is big-endian (most significant byte first), reflecting its Motorola 68k roots on early Macs. The conversion with pcm_s16be handles this at the sample level. One important limitation: this tool converts WAV files containing standard PCM codecs (pcm_s16le, pcm_s24le, pcm_s32le, pcm_f32le), but WAV files encoded with compressed codecs like ADPCM or MP3 inside a WAV container will be decoded to PCM first, then written as pcm_s16be AIFF — which may alter the audio characteristics of those specific files. AIFF does not support the FLAC codec that WAV can technically carry. Additionally, AIFF has no native support for broadcast extension metadata (bext chunks) found in BWF-flavored WAV files used in professional broadcast workflows; that metadata will be lost. For most studio and consumer use cases, the output AIFF is a fully interchangeable, high-fidelity replacement for the original WAV.