Convert AAC to AIFF — Free Online Tool
Convert AAC audio files to AIFF format by decoding the lossy AAC stream and re-encoding it as uncompressed PCM audio (16-bit big-endian) — the native lossless format used by macOS and professional Apple audio workflows. This is ideal when you need to bring compressed AAC audio into a DAW or audio editing environment that requires uncompressed source files.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AAC file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
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, meaning the original audio data was permanently altered when the AAC file was first created — some audio information was discarded to achieve smaller file sizes. During this conversion, FFmpeg fully decodes the AAC bitstream back into raw PCM audio samples, then encodes those samples into the AIFF container using the pcm_s16be codec (16-bit signed big-endian PCM). Because AAC is lossy, this conversion does not recover any of the audio detail that was lost during the original AAC encoding — what you get is a lossless container holding the exact decoded output of the AAC file, with no further quality degradation from this conversion step itself. The resulting AIFF file will be significantly larger than the AAC source, as uncompressed 16-bit stereo audio at 44.1kHz requires roughly 10MB per minute, compared to a few MB per minute for AAC.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. This conversion runs entirely in your browser via FFmpeg.wasm, a WebAssembly port of FFmpeg — no files leave your device. |
-i input.aac
|
Specifies the input AAC audio file. FFmpeg will detect whether this is a raw ADTS AAC stream or an AAC file wrapped in an M4A/MPEG-4 container and decode it accordingly. |
-c:a pcm_s16be
|
Sets the audio codec for the output to PCM signed 16-bit big-endian — the standard uncompressed audio encoding used in Apple AIFF files. This decodes the lossy AAC audio and re-stores the samples as raw, uncompressed 16-bit integers in the byte order AIFF requires. |
output.aiff
|
Defines the output filename with the .aiff extension, which tells FFmpeg to wrap the uncompressed PCM audio in an Audio Interchange File Format (AIFF) container — Apple's native lossless audio format compatible with macOS, Logic Pro, and professional audio tools. |
Common Use Cases
- Importing purchased iTunes or Apple Music downloads (distributed as AAC) into a DAW like Logic Pro or Pro Tools, which often works more reliably with uncompressed AIFF source files
- Preparing AAC audio files for use in Final Cut Pro or other Apple-native video editing tools that prefer AIFF for timeline audio
- Archiving AAC recordings from an iPhone or iPad into AIFF format for long-term storage on a Mac, where AIFF is the standard uncompressed format
- Converting AAC podcast episodes or interview recordings to AIFF before applying audio restoration or mastering effects in a professional audio editor
- Delivering audio assets to a mastering engineer or post-production studio that requires uncompressed AIFF stems rather than compressed AAC files
- Batch-converting an AAC music library to AIFF for use with audio hardware or software that does not support AAC decoding natively
Frequently Asked Questions
No. AAC is a lossy format, which means audio data was permanently removed when the file was originally encoded to AAC. Converting to AIFF produces an uncompressed file that faithfully represents what the AAC decoder outputs, but it cannot restore the discarded information from the original lossy compression. The AIFF file will be bit-for-bit identical in audio content to the decoded AAC — just uncompressed and in a lossless container. If quality is a concern, always start from the highest-quality source available.
AAC uses perceptual audio compression to reduce file size, typically achieving 8:1 to 12:1 compression ratios compared to uncompressed audio. AIFF with pcm_s16be stores every audio sample without any compression — a standard stereo 44.1kHz 16-bit AIFF file uses approximately 10MB per minute. A 5MB AAC file might expand to 50–70MB as AIFF. This size increase is expected and does not indicate a problem with the conversion.
pcm_s16be stands for PCM signed 16-bit big-endian — it stores each audio sample as a 16-bit integer in big-endian byte order, which is the native byte order Apple uses in AIFF files. 16-bit is the standard CD-quality bit depth and is appropriate for most AAC sources, since consumer AAC files are rarely mastered at more than 16-bit equivalent quality. If you need higher bit depth for professional workflows, you can modify the FFmpeg command to use pcm_s24be (24-bit) or pcm_s32be (32-bit) instead of pcm_s16be.
AIFF has limited metadata support compared to AAC's M4A container. FFmpeg will attempt to copy basic ID3-style tags (title, artist, album) into the AIFF file during conversion, but embedded album artwork is typically not preserved in AIFF files. If metadata preservation is critical, you may want to use a dedicated tagging tool after conversion to re-apply artwork and extended tags to the AIFF file.
Replace pcm_s16be in the command with another PCM codec to change the bit depth. Use pcm_s24be for 24-bit audio (common in professional studio work), 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.aiff. Note that changing the bit depth of the AIFF output does not recover any quality lost during the original AAC encoding — the source audio is still limited by the fidelity of the decoded AAC stream.
Yes. On macOS or Linux, you can use a shell loop to process multiple files: for f in *.aac; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.aac}.aiff"; done. On Windows Command Prompt, use: for %f in (*.aac) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This is particularly useful when converting a large library of AAC files — the in-browser tool handles one file at a time, while the local FFmpeg command can be scripted for bulk processing.
Technical Notes
AAC audio is typically stored in either an M4A container (MPEG-4 Audio) or a raw ADTS stream (.aac extension). FFmpeg handles both transparently during this conversion. The output AIFF file uses pcm_s16be, which encodes samples as 16-bit signed integers in big-endian byte order — this is the historical Apple standard and is universally compatible with macOS audio applications, Logic Pro, GarageBand, and hardware samplers that read AIFF. One important limitation: AAC supports multi-channel audio (up to 5.1 or 7.1), and AIFF technically supports multi-channel PCM as well, but compatibility with multi-channel AIFF varies across applications. For stereo AAC sources, this conversion is entirely straightforward. The sample rate of the source AAC is preserved in the output AIFF without resampling — if your AAC was encoded at 44.1kHz, the AIFF will also be 44.1kHz. If you require a specific sample rate for your workflow (e.g., 48kHz for video post-production), add the -ar 48000 flag to the command. Because AAC uses a codec-specific delay (the 'encoder delay' or 'priming samples' added by AAC encoders), FFmpeg applies the correct gapless playback correction when decoding, so the AIFF output should be cleanly trimmed without leading silence artifacts.