Convert AIFF to M4A — Free Online Tool
Convert AIFF audio files to M4A format by transcoding Apple's uncompressed PCM audio into AAC, the lossy codec used by iTunes and Apple Music. This is the standard workflow for reducing the large file sizes of studio-quality AIFF recordings while keeping them in Apple's native ecosystem.
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 stores audio as raw, uncompressed PCM data (typically 16-bit or 24-bit big-endian samples), which means every sample is stored at full resolution with no compression — a 3-minute stereo track at 16-bit/44.1kHz takes roughly 30MB. During this conversion, FFmpeg decodes those raw PCM samples and re-encodes them using the AAC (Advanced Audio Coding) codec, which applies perceptual compression to discard audio data that human hearing is unlikely to notice. The resulting audio is then wrapped in an MPEG-4 (.m4a) container. Because AAC is inherently lossy, this is a one-way transformation — the original uncompressed data cannot be recovered from the output M4A file. At the default bitrate of 128k, the output file will be roughly 10–15× smaller than the source AIFF.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm), so no files leave your device. |
-i input.aiff
|
Specifies the input file — an AIFF file containing uncompressed PCM audio (commonly 16-bit or 24-bit big-endian samples). FFmpeg auto-detects the AIFF container and its PCM codec from the file header. |
-c:a aac
|
Sets the audio codec for the output to AAC (Advanced Audio Coding), the lossy compression format native to Apple's iTunes ecosystem and used by Apple Music. This is what transforms the raw PCM samples from the AIFF into a compact, perceptually compressed audio stream. |
-b:a 128k
|
Sets the AAC audio bitrate to 128 kilobits per second — a widely accepted balance between file size and perceived audio quality for music and voice content. Increase this to 192k or 256k for higher fidelity, or lower it to 96k for speech-only content where smaller size matters more. |
-vn
|
Explicitly disables video output, ensuring the M4A file is a clean audio-only container. This prevents any unexpected or spurious stream data from the input being written as a video track, which would break compatibility with iTunes and Apple devices. |
output.m4a
|
The output filename with the .m4a extension, which tells FFmpeg to wrap the AAC audio in an MPEG-4 audio container. The .m4a extension specifically signals to Apple software that this is an audio-only file, enabling correct handling in iTunes, QuickTime, Finder, and iOS. |
Common Use Cases
- Preparing high-resolution AIFF masters from a recording session for distribution on iTunes or Apple Music, where M4A/AAC is the native delivery format
- Reducing the storage footprint of a large AIFF sample library on an iPhone or iPad, where device storage is limited and lossless files are impractical
- Converting AIFF podcast recordings exported from Logic Pro or GarageBand into M4A for upload to podcast hosting platforms that favor compressed audio
- Sharing AIFF audio proofs with clients or collaborators who need smaller files that play directly in Safari, QuickTime, or Apple's built-in media players without any additional software
- Archiving a collection of AIFF CD rips into space-efficient M4A files for everyday listening while keeping the original lossless AIFFs stored separately as masters
- Encoding AIFF voice-over recordings into M4A at a controlled bitrate for use in iMovie, Final Cut Pro, or Keynote presentations where file size affects project portability
Frequently Asked Questions
Yes — this conversion is lossy, so some audio information is permanently discarded. However, AAC at 128k is generally considered transparent (indistinguishable from the source) for most listeners on typical playback equipment. If you are doing critical listening or mastering work, consider using 192k or 256k for a more conservative quality tradeoff. The difference is most audible on complex high-frequency content like cymbals or acoustic guitar at lower bitrates.
Not when using AAC encoding. AAC is a lossy codec that operates on a frequency-domain model rather than preserving individual PCM bit depths, so the 24-bit or 32-bit precision of your AIFF is not carried over into the encoded AAC stream. If you need to preserve bit depth inside an M4A container, you could encode to FLAC (which M4A technically supports), but that file may not play in all Apple applications. For lossless preservation with Apple compatibility, Apple Lossless (ALAC) in an M4A container is a better option.
Change the -b:a 128k flag to a higher bitrate such as -b:a 192k or -b:a 256k. For example: ffmpeg -i input.aiff -c:a aac -b:a 256k -vn output.m4a. AAC at 256k is considered high quality and is the bitrate Apple Music uses for its standard streams. Going beyond 320k yields diminishing returns with AAC and is rarely worthwhile.
FFmpeg will attempt to copy standard metadata tags automatically during conversion. AIFF stores metadata in ID3 or AIFF chunk format, while M4A uses iTunes-style atom tags (such as ©nam, ©ART). FFmpeg maps common fields like title, artist, album, and year between these formats, but less common or custom tags may not survive the conversion. You may want to verify tags in iTunes or a tag editor like MP3Tag after converting.
The -vn flag explicitly tells FFmpeg to ignore any video streams in the input and write no video to the output. While standard AIFF files never contain video, some edge-case or malformed files might contain unexpected stream data. Including -vn also ensures the MPEG-4 output is a clean audio-only M4A file rather than an MP4 with a blank or corrupt video track, which could confuse iTunes or other players.
Yes. On macOS or Linux, you can loop over all AIFF files in a directory with a shell one-liner: for f in *.aiff; do ffmpeg -i "$f" -c:a aac -b:a 128k -vn "${f%.aiff}.m4a"; done. On Windows PowerShell, use: Get-ChildItem *.aiff | ForEach-Object { ffmpeg -i $_.FullName -c:a aac -b:a 128k -vn ($_.BaseName + '.m4a') }. The browser-based tool on this page processes one file at a time, so the FFmpeg command is particularly useful for bulk conversions.
Technical Notes
AIFF uses big-endian PCM encoding (pcm_s16be, pcm_s24be, etc.), which FFmpeg decodes natively before passing samples to the AAC encoder. The default AAC encoder in FFmpeg (aac) is a native implementation that produces good quality at 128k and above; for even higher quality, the external libfdk_aac encoder is generally preferred when available in a custom FFmpeg build, though it is not included in most standard distributions due to licensing. The M4A container is technically an MPEG-4 Part 14 file with a .m4a extension and is identical in structure to .mp4 — the extension change signals to iTunes and Apple devices that the file is audio-only. M4A supports chapter markers, which AIFF does not, so if you add chapters after conversion they will be preserved in the container. One known limitation is gapless playback metadata: AIFF files used in gapless album sequences (common with CD rips) may not have their gapless iTunSMPB tags correctly written to the M4A output by FFmpeg, which can introduce a small gap between tracks in iTunes. File size reduction from AIFF to M4A at 128k is dramatic — a 50MB AIFF will typically produce a 3–4MB M4A, making this conversion essential for mobile distribution.