Convert AAC to AIF — Free Online Tool
Convert AAC audio files to AIF (Audio Interchange File Format), decoding the lossy AAC stream and re-encoding it as uncompressed PCM audio (16-bit big-endian) for high-fidelity storage on Mac systems. Ideal when you need an uncompressed master file from an AAC source for use in Apple-ecosystem audio workflows.
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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
AAC is a lossy compressed format — it discards audio data during encoding to achieve small file sizes. When converting to AIF, FFmpeg fully decodes the AAC bitstream back into raw PCM audio samples, then writes them into an AIFF container using the pcm_s16be codec (16-bit signed big-endian PCM). This is a full decode-and-re-encode operation, not a remux. Importantly, because AAC already discarded audio information during its original compression, the resulting AIF file will be uncompressed but will not recover any quality lost when the AAC was first created — it will simply preserve the decoded audio at full resolution going forward. The output file will be significantly larger than the AAC source, reflecting the uncompressed nature of AIFF.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop command line. |
-i input.aac
|
Specifies the input file — in this case, an AAC audio file. FFmpeg detects the AAC codec and demuxes the compressed audio stream in preparation for decoding. |
-c:a pcm_s16be
|
Sets the audio codec for the output to pcm_s16be — 16-bit signed big-endian PCM — which is the standard uncompressed audio encoding used inside an AIFF container and natively compatible with Apple software. |
output.aif
|
Defines the output filename with the .aif extension, which tells FFmpeg to wrap the uncompressed PCM audio in an AIFF container — Apple's native uncompressed audio format for Mac-based audio production. |
Common Use Cases
- Importing purchased iTunes or Apple Music downloads (in AAC format) into a professional DAW like Logic Pro or Pro Tools, which prefer or require uncompressed AIFF masters for multi-track editing sessions.
- Archiving AAC audio files as uncompressed AIF for long-term storage in an Apple-centric media library, avoiding repeated lossy decode-encode cycles in future workflows.
- Preparing AAC podcast episodes or voice recordings for professional post-production, where the AIF format integrates cleanly with Final Cut Pro and other Apple software without transcoding delays.
- Converting AAC audio from a video export (e.g., stripped from an MP4 file) into AIF so it can be handed off to a mastering engineer or sound designer working on a Mac.
- Creating uncompressed AIF versions of AAC sound effects or music beds for use in Xcode-based iOS or macOS app development, where AIFF is a natively supported audio asset format.
- Standardizing a mixed collection of AAC files to a single uncompressed AIF format for consistent playback and editing across a professional Apple audio production pipeline.
Frequently Asked Questions
No — converting from AAC to AIF will not recover any audio quality that was lost during the original AAC encoding. AAC is a lossy format, meaning it permanently discards certain audio information to compress the file. The AIF output will be an uncompressed representation of the decoded AAC audio, which means it will be bit-for-bit faithful to what the AAC file contained, but it will not restore frequencies or detail that AAC's perceptual encoding removed. Think of it as 'lossless preservation of a lossy source' rather than a quality upgrade.
AAC compresses audio aggressively — a 128k AAC file might be 10 to 15 times smaller than its uncompressed equivalent. AIFF stores raw, uncompressed PCM samples, so the file size is determined purely by sample rate, bit depth, number of channels, and duration. For example, a 3-minute stereo AAC file at 128k might be around 3MB, while the equivalent uncompressed 16-bit AIF would be approximately 30–32MB. This is expected and is the trade-off of moving from compressed to uncompressed audio.
pcm_s16be stands for PCM (Pulse-Code Modulation) signed 16-bit big-endian — the 's16' means 16-bit signed integers are used to represent each audio sample, and 'be' means the byte order is big-endian, which is the native byte order for Apple's AIFF format. 16-bit PCM is the same depth used in standard CD audio and is perfectly suitable for most professional workflows. However, if you need 24-bit depth for audio mastering or post-production, you can modify the FFmpeg command to use pcm_s24be instead.
AAC files (typically in an M4A/MP4 container) can carry rich iTunes-style metadata, but AIFF has more limited and differently structured metadata support. FFmpeg will attempt to map common tags such as title and artist, but not all AAC metadata fields have direct AIFF equivalents, and some tags may be dropped or not recognized by all applications. If metadata preservation is critical, verify the output in your target application (e.g., iTunes or Logic Pro) after conversion.
By default, the command uses pcm_s16be for 16-bit output. To get a higher bit depth, replace pcm_s16be with pcm_s24be for 24-bit, pcm_s32be for 32-bit integer, or pcm_f32be for 32-bit floating-point. For example: ffmpeg -i input.aac -c:a pcm_s24be output.aif. Since the source is AAC (typically encoded at 16-bit equivalent perceptual quality), you are unlikely to hear a difference above 16-bit, but 24-bit output is common practice in professional audio pipelines to maintain headroom for further processing.
The single-file command shown here must be run once per file, but you can automate batch conversion with a simple shell script. On macOS or Linux, you can use: for f in *.aac; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.aac}.aif"; done. On Windows PowerShell: Get-ChildItem *.aac | ForEach-Object { ffmpeg -i $_.FullName -c:a pcm_s16be ($_.BaseName + '.aif') }. This is especially useful for large collections since the browser tool handles one file at a time, while the desktop FFmpeg command has no such limitation.
Technical Notes
The AAC-to-AIF conversion is a full decode operation: FFmpeg uses its native AAC decoder to reconstruct PCM audio from the compressed bitstream, then writes it into an AIFF container. No audio information beyond what AAC preserved is recoverable. The default output codec is pcm_s16be (16-bit big-endian PCM), which matches the canonical AIFF specification and is broadly compatible with all Apple software including Logic Pro, GarageBand, Final Cut Pro, and QuickTime Player. AIFF does not support multiple audio tracks, subtitles, chapters, or video streams, so any of those elements present in the source would be ignored. The AIFF container uses big-endian byte ordering (a legacy of Motorola 68k and PowerPC architecture), which distinguishes it from the nearly identical AIFF-C (AIFC) and WAV formats. If you need little-endian PCM (e.g., for WAV compatibility), WAV would be the more appropriate output target. Sample rate is preserved from the AAC source — common values are 44.1 kHz or 48 kHz — and is not altered by this command. For AAC files sourced from Apple's FairPlay DRM-protected ecosystem (e.g., older iTunes purchases), FFmpeg will be unable to decode the file, as DRM removal is outside FFmpeg's scope.