Convert AIF to WMA — Free Online Tool
Convert AIF files to WMA by transcoding Apple's uncompressed PCM audio into Microsoft's wmav2 codec at 128kbps. This is useful for making high-fidelity Mac audio files compatible with Windows Media Player, Xbox, and other Microsoft ecosystem devices that don't natively support AIF.
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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
AIF stores audio as raw uncompressed PCM data (typically pcm_s16be — 16-bit big-endian samples), which means every audio sample is stored at full resolution with no compression. During this conversion, FFmpeg decodes those raw PCM samples and re-encodes them using the wmav2 codec, a lossy compression algorithm that applies perceptual audio modeling to discard frequencies less audible to the human ear. The result is a dramatically smaller WMA file. Because this is a lossy encode from a lossless source, some audio detail is permanently discarded — but the original AIF file remains untouched, so you always have the lossless master to re-convert at any quality setting.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application — the open-source multimedia processing engine that handles decoding the AIF input and encoding the WMA output. This tool runs FFmpeg compiled to WebAssembly (ffmpeg.wasm) directly in your browser, so no files leave your device. |
-i input.aif
|
Specifies the input file — an AIF audio file containing uncompressed PCM audio data in Apple's Audio Interchange File Format. FFmpeg reads the PCM samples and decodes the big-endian byte order for further processing. |
-c:a wmav2
|
Sets the audio codec to wmav2 (Windows Media Audio v2), Microsoft's second-generation lossy audio encoder. This is the standard codec for WMA files and is supported by all modern Windows Media Player versions, Xbox, and Windows-native media software. |
-b:a 128k
|
Sets the audio bitrate to 128 kilobits per second for the wmav2 encoder. This is the conventional balance point between file size and audio quality for WMA — roughly one-tenth the data rate of the uncompressed 16-bit 44.1kHz AIF source, at a quality level suitable for music and general listening. |
output.wma
|
Specifies the output filename with the .wma extension, which tells FFmpeg to wrap the wmav2-encoded audio stream in an ASF (Advanced Systems Format) container — the standard container for Windows Media Audio files. |
Common Use Cases
- Sharing Mac-produced audio recordings with colleagues or clients who use Windows Media Player or other Microsoft-native applications that don't support AIF
- Reducing the file size of large uncompressed AIF audio exports from Logic Pro or GarageBand before distributing them over email or file-sharing services
- Preparing voice-over or podcast recordings originally captured in AIF format for upload to platforms or CMS systems that require WMA input
- Converting AIF music files or sound effects for use in older Windows-based multimedia projects, kiosks, or presentations built around Windows Media
- Archiving a compressed WMA copy of an AIF recording for everyday listening while keeping the original lossless AIF as a production master
- Making AIF audio files playable on Xbox consoles or Windows Phone devices that support WMA but not the AIF container
Frequently Asked Questions
Yes — this conversion is lossy. AIF stores audio as uncompressed PCM, meaning every bit of the original recording is preserved. WMA uses the wmav2 codec, which applies perceptual compression to reduce file size, permanently discarding some audio data in the process. At the default 128kbps bitrate, the result is generally transparent for casual listening, but a trained ear may detect differences on critical material. If quality is paramount, keep the original AIF as your master.
Significantly smaller. A 16-bit stereo AIF file at 44.1kHz consumes roughly 10MB per minute of audio. At 128kbps WMA, the same minute of audio is about 1MB — a roughly 10:1 reduction. The exact ratio depends on the sample rate and bit depth of your source AIF, but WMA compression consistently produces far smaller files than uncompressed PCM at audibly acceptable quality levels.
wmav2 is the second-generation Windows Media Audio codec and produces better audio quality than wmav1 at the same bitrate by using more sophisticated encoding techniques. wmav1 is an older codec with broader compatibility for very early Windows Media Player versions (pre-WMP 7), but wmav2 has been universally supported since Windows XP and is the appropriate choice for virtually all modern use cases.
Possibly, but not guaranteed. AIF stores metadata in AIFF chunk structures, while WMA uses ASF (Advanced Systems Format) metadata tags. FFmpeg will attempt to map common fields like title, artist, and album during conversion, but custom or non-standard AIF metadata chunks may be dropped. You should verify the output WMA file's tags in a tool like Windows Media Player or MediaInfo after conversion.
Replace the '128k' value in the '-b:a 128k' flag with your desired bitrate. Supported options for wmav2 include 64k, 96k, 128k, 160k, 192k, 256k, and 320k. For example, to get higher quality output, use 'ffmpeg -i input.aif -c:a wmav2 -b:a 192k output.wma'. Higher bitrates produce larger files that more closely approximate the lossless AIF source, while lower bitrates like 64k are suitable for speech or voice-only content.
Yes, using a shell loop. On Linux or macOS, run: 'for f in *.aif; do ffmpeg -i "$f" -c:a wmav2 -b:a 128k "${f%.aif}.wma"; done'. On Windows Command Prompt, use: 'for %f in (*.aif) do ffmpeg -i "%f" -c:a wmav2 -b:a 128k "%~nf.wma"'. The browser-based tool on this page processes one file at a time, so the FFmpeg command is the recommended approach for bulk conversions.
Technical Notes
AIF uses big-endian PCM encoding (hence codec names like pcm_s16be for 16-bit signed big-endian), which is the native byte order of classic Motorola and PowerPC Mac hardware. FFmpeg handles the byte-order conversion transparently during decoding before feeding samples to the wmav2 encoder. The WMA container is based on Microsoft's ASF (Advanced Systems Format), which supports metadata, DRM, and streaming — features the AIF format lacks entirely. One important limitation: WMA and ASF do not support multi-channel audio beyond stereo in wmav2 mode (surround audio requires WMA Pro, a different codec). If your source AIF is a multi-channel surround file, the conversion may downmix channels or fail, and you should explicitly specify channel mapping with '-ac 2' to force stereo output. Also note that because AIF is a lossless format, no amount of WMA bitrate can recover quality beyond the original PCM signal — 320kbps WMA from a 16-bit AIF source will not sound better than 128kbps in any meaningful way, so the default 128kbps is a practical choice for most use cases.