Convert AIFF to AAC — Free Online Tool
Convert AIFF audio files to AAC format using FFmpeg's built-in AAC encoder, compressing Apple's uncompressed PCM audio (typically pcm_s16be or pcm_s24be) into a streamable, widely-compatible lossy format at 128k bitrate by default. Ideal for reducing large AIFF masters for distribution on streaming platforms, iTunes, or mobile devices without needing Pro Tools or Logic Pro.
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 samples — no encoding step is needed to decode it, so FFmpeg reads the raw audio data directly. During conversion, FFmpeg transcodes those PCM samples into AAC using its native AAC encoder, applying perceptual audio coding to discard sounds the human ear is least likely to notice, dramatically reducing file size. Because AIFF carries no video or subtitle streams, the process is purely an audio transcode — there is no remuxing or stream copying involved. The output is a raw AAC bitstream wrapped in a .aac container, which is distinct from the .m4a container (MPEG-4 Audio) but uses the same underlying AAC codec.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine running under the hood, compiled to WebAssembly (FFmpeg.wasm) to execute entirely inside your browser without any server upload. |
-i input.aiff
|
Specifies the input file as an AIFF audio file. FFmpeg detects the AIFF container and its PCM audio codec (e.g., pcm_s16be or pcm_s24be) automatically, preparing the raw uncompressed samples for transcoding. |
-c:a aac
|
Selects FFmpeg's built-in AAC encoder to transcode the uncompressed PCM audio from the AIFF source into AAC — a lossy, perceptually-coded format optimized for streaming and mobile playback. |
-b:a 128k
|
Sets the AAC audio bitrate to 128 kilobits per second, a widely-used default that balances file size and audio quality well for stereo music and voice content. Increase to 192k or 256k for higher fidelity, or lower to 96k for voice-only recordings. |
output.aac
|
Defines the output filename and tells FFmpeg to write a raw AAC bitstream (.aac). This format is distinct from .m4a, which wraps AAC in an MPEG-4 container with richer metadata and broader iTunes compatibility. |
Common Use Cases
- Preparing high-resolution AIFF masters from a recording session for upload to Apple Music, Spotify, or other streaming services that accept AAC
- Reducing the file size of large AIFF sound libraries (e.g., sample packs or foley recordings) for distribution or archiving without maintaining lossless originals
- Converting AIFF voice-over or podcast recordings exported from Logic Pro or GarageBand into a lightweight AAC file for web embedding or podcast hosting
- Creating AAC previews or low-bandwidth versions of uncompressed AIFF audio for client review without sending multi-hundred-megabyte files
- Encoding AIFF audio assets from a game or film project into AAC for use in iOS or Android apps, where AAC is the natively preferred audio format
- Batch-converting a AIFF music collection ripped from CDs on macOS into AAC for playback on devices with limited storage
Frequently Asked Questions
AIFF stores audio as uncompressed PCM, so a 3-minute stereo track at 16-bit/44.1kHz occupies roughly 30–35MB. The same audio encoded to AAC at 128k bitrate typically lands around 2.8–3MB — a reduction of approximately 90%. At higher AIFF bit depths (24-bit or 32-bit), which are common for professional recordings, the size difference is even more dramatic since the source file is larger but the AAC output stays proportional to its bitrate setting.
Yes — AAC is a lossy format, so some audio information is permanently discarded during encoding. However, at 128k or higher, most listeners cannot distinguish AAC from the lossless AIFF source in a blind test. The key point is that you should always keep your original AIFF as a master; once converted to AAC, you cannot recover the lost detail by converting back. If you need lossless compression for archiving instead of distribution, consider FLAC or Apple Lossless (ALAC) as alternatives.
Both use the AAC audio codec, but they differ in container format. A .aac file is a raw AAC bitstream with minimal container overhead, while .m4a wraps AAC inside an MPEG-4 container, which supports richer metadata (album art, track numbers, lyrics) and is more broadly compatible with iTunes, iPhone, and Apple ecosystem apps. If you need iTunes or Music app compatibility, consider targeting .m4a instead. The FFmpeg command for .m4a would be nearly identical — just change the output filename to output.m4a.
AIFF supports basic ID3-style metadata such as title, artist, and album, and FFmpeg will attempt to map these tags to the AAC output by default. However, because a raw .aac container has limited metadata support compared to .m4a, some tags may not persist or may not be readable by all players. If metadata preservation is important, wrapping the AAC audio in an .m4a container is strongly recommended, as it uses a more capable metadata structure.
Replace the 128k value in the -b:a 128k flag with your desired bitrate. For example, use -b:a 192k for higher quality suitable for music, -b:a 96k for spoken word or podcasts where bandwidth is a priority, or -b:a 256k for near-transparent quality on critical listening content. The full command at 192k would be: ffmpeg -i input.aiff -c:a aac -b:a 192k output.aac. AAC generally achieves transparent quality at 128–192k for most stereo content.
Yes. On Linux or macOS, you can use a shell loop: for f in *.aiff; do ffmpeg -i "$f" -c:a aac -b:a 128k "${f%.aiff}.aac"; done. On Windows Command Prompt, use: for %f in (*.aiff) do ffmpeg -i "%f" -c:a aac -b:a 128k "%~nf.aac". This processes every AIFF file in the current directory and outputs a matching AAC file, preserving the base filename. This approach is particularly useful for converting large sample libraries or full album directories in one step.
Technical Notes
AIFF files can contain PCM audio at various bit depths — pcm_s16be (CD quality), pcm_s24be (studio standard), pcm_s32be, and floating-point variants (pcm_f32be, pcm_f64be). FFmpeg handles all of these transparently when encoding to AAC, downsampling the bit depth internally as part of the transcode. The built-in FFmpeg AAC encoder (-c:a aac) is a solid general-purpose encoder, but if your FFmpeg build includes libfdk_aac (common in third-party builds), substituting -c:a libfdk_aac typically yields better perceptual quality at equivalent bitrates, particularly below 128k. The output .aac file is a raw ADTS-framed AAC bitstream — it lacks the seekable index and full metadata support of an MP4 container. For files intended for Apple platforms, the .m4a extension with an MPEG-4 container is preferred. Additionally, if your AIFF source is multichannel (e.g., 5.1 surround), be aware that the AAC encoder may require explicit channel layout flags, as default behavior can vary across FFmpeg versions.