Convert AIFF to WAV — Free Online Tool
Convert AIFF audio files to WAV format directly in your browser, transcoding Apple's big-endian PCM audio (pcm_s16be) to the little-endian PCM format (pcm_s16le) that Windows, Linux, and virtually every DAW and media tool expects. Both formats are lossless and uncompressed, so you lose nothing in audio quality during conversion.
to
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
AIFF and WAV are both uncompressed PCM audio containers, but they differ in byte order and origin: AIFF uses big-endian byte ordering (a legacy of Motorola/Apple hardware), while WAV uses little-endian byte ordering (standard on x86 Windows and Linux systems). This conversion remuxes the raw audio samples from their AIFF container into a WAV container and simultaneously flips the byte order of each PCM sample from big-endian (pcm_s16be) to little-endian (pcm_s16le). Because both formats store audio at the same bit depth and sample rate with no compression, no audio data is discarded or degraded — the waveform is preserved sample-for-sample. The output WAV file will be nearly identical in file size to the input AIFF.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, which handles the reading, transcoding, and writing of the audio file. In the browser-based version of this tool, FFmpeg runs as a WebAssembly (wasm) binary — no installation needed and no files leave your device. |
-i input.aiff
|
Specifies the input file, an AIFF audio file containing raw PCM audio encoded in big-endian byte order (typically pcm_s16be). FFmpeg reads the AIFF container header to determine the sample rate, bit depth, channel count, and codec before decoding. |
-c:a pcm_s16le
|
Sets the output audio codec to 16-bit signed PCM in little-endian byte order, which is the standard uncompressed audio format for WAV files on Windows and Linux systems. This performs the critical byte-order conversion from AIFF's big-endian samples to WAV's little-endian samples while keeping all audio data intact. |
output.wav
|
Specifies the output filename with a .wav extension, which tells FFmpeg to write the converted audio into a WAV (RIFF) container. The WAV container wraps the pcm_s16le audio stream with a standard RIFF header, making it immediately compatible with virtually all media players, DAWs, and audio tools on any platform. |
Common Use Cases
- Importing Apple Logic Pro or GarageBand audio exports into Windows-based DAWs like FL Studio or Ableton Live, which expect WAV files over AIFF
- Preparing mastered audio tracks for distribution platforms or CD replication services that mandate WAV format delivery
- Making AIFF sound effects or sample packs compatible with Windows audio production software, games engines like Unity or Unreal, or sound design tools that don't support AIFF
- Converting AIFF field recordings from a Mac-based audio recorder for use in video editing on a Windows machine with DaVinci Resolve or Premiere Pro
- Uploading audio to broadcast or podcast platforms that accept WAV but not AIFF, such as certain radio automation systems or streaming ingestion tools
- Archiving or sharing audio files in a format with broader OS and device compatibility, since WAV is universally supported while AIFF support is less consistent on non-Apple platforms
Frequently Asked Questions
No. Both AIFF and WAV store audio as raw, uncompressed PCM data, so there is no encoding or decoding step that would introduce loss. The conversion simply repackages the same audio samples into a different container and reverses the byte order from big-endian to little-endian. The waveform, bit depth, and sample rate remain exactly the same.
Because both AIFF and WAV are uncompressed audio formats. The audio payload — the raw PCM samples — is the same size in both containers. The only size difference comes from slight differences in header and metadata structure, which amounts to a few hundred bytes at most. Unlike converting to MP3 or AAC, no compression is applied during this conversion.
Not reliably. AIFF stores metadata in its ANNO and ID3 chunks, while WAV uses LIST/INFO chunks or an embedded ID3 tag. FFmpeg does attempt to map common metadata fields, but some tags may be dropped or not recognized by all WAV-reading applications. If preserving detailed metadata is important, you should verify the output tags with a tool like MediaInfo or a tag editor after conversion.
Both pcm_s16be and pcm_s16le are 16-bit signed PCM audio — identical in quality and data — but they differ in how each 16-bit sample is stored in memory. 'be' stands for big-endian (most significant byte first, used by AIFF on older Apple/Motorola hardware) and 'le' stands for little-endian (least significant byte first, standard on modern x86 processors and Windows). Playing a big-endian file on a little-endian system without conversion produces loud crackling noise or silence, which is why the byte-swap matters.
Replace pcm_s16le in the command with pcm_s24le for 24-bit output or pcm_s32le for 32-bit output: ffmpeg -i input.aiff -c:a pcm_s24le output.wav. You should match the output bit depth to your source AIFF's bit depth — if the original was recorded at 24-bit, converting to pcm_s16le would reduce bit depth and is the one scenario where audio data would be truncated. Use ffprobe input.aiff to check the original bit depth before converting.
Yes. On macOS or Linux, you can run a shell loop: for f in *.aiff; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.aiff}.wav"; done. On Windows Command Prompt, use: for %f in (*.aiff) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This applies the same pcm_s16le transcoding to every AIFF file in the current directory and outputs a matching WAV file for each.
Technical Notes
AIFF (Audio Interchange File Format) and WAV (Waveform Audio File Format) are both IFF-derived container formats for uncompressed PCM audio, though AIFF descends from EA IFF 85 on Apple/Motorola platforms and WAV descends from RIFF on Microsoft/Intel platforms — which is the direct cause of their opposing byte orders. This converter uses FFmpeg's pcm_s16be decoder for the input and pcm_s16le encoder for the output, performing an in-memory byte-swap on each sample with no quality loss. Both formats support the same sample rates (8kHz through 192kHz) and the same bit depths (8, 16, 24, 32-bit), so no resampling or requantization occurs unless you explicitly request it. One known limitation is that AIFF-C (the compressed variant of AIFF, sometimes using GSM or MACE codecs) may require different handling, though plain AIFF with pcm_s16be is by far the most common variant encountered in practice. Loop point and marker metadata stored in AIFF's MARK and INST chunks — commonly used in samplers — will not be preserved in the WAV output, as WAV uses a different mechanism (smpl chunk) for this data, and FFmpeg does not currently map between them.