Convert AAC to AIFC — Free Online Tool
Convert AAC audio files to AIFC format in your browser, transcoding the lossy AAC stream to uncompressed 16-bit big-endian PCM audio (pcm_s16be) — the lossless PCM codec native to Apple's AIFF/AIFC container. This is ideal for bringing compressed AAC audio into professional audio workflows that require uncompressed source material.
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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 audio uses perceptual lossy compression, discarding audio data the human ear is less likely to notice to achieve small file sizes. During this conversion, FFmpeg fully decodes the AAC bitstream back to raw PCM audio samples, then re-encodes those samples as 16-bit signed big-endian PCM and wraps them in an AIFC container. Because AAC is lossy, the artifacts introduced during the original AAC encoding cannot be recovered — the resulting AIFC file contains a lossless representation of the decoded AAC audio, not the original pre-compression signal. The AIFC container is Apple's extension of AIFF that formally supports both uncompressed and compressed audio chunk types; in this case we use uncompressed PCM, resulting in a significantly larger file than the source AAC.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application, the open-source multimedia framework that handles all decoding, encoding, and container muxing in this conversion. In the browser, this runs via FFmpeg.wasm compiled to WebAssembly. |
-i input.aac
|
Specifies the input file — an AAC audio file, which FFmpeg will decode from its lossy compressed bitstream into raw PCM audio samples in memory before re-encoding for the AIFC output. |
-c:a pcm_s16be
|
Sets the audio codec for the output to signed 16-bit big-endian PCM, the native uncompressed audio format of Apple's AIFF/AIFC container. This transforms the lossy AAC stream into a lossless, fully uncompressed representation of the decoded audio. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For pcm_s16be, the actual bitrate is mathematically determined by sample rate and bit depth rather than this parameter, so its effect is largely nominal — the PCM stream will be written at its natural fixed bitrate regardless. |
output.aifc
|
Defines the output filename and signals FFmpeg to write an AIFC container. FFmpeg uses the .aifc file extension to select the AIFF/AIFC muxer, which wraps the pcm_s16be audio stream in Apple's big-endian audio interchange format. |
Common Use Cases
- Importing AAC podcast recordings or voice memos into professional DAWs like Logic Pro or Pro Tools, which prefer uncompressed AIFF/AIFC source files for editing
- Preparing AAC-encoded music downloaded from iTunes for use in a sample library or audio production project that mandates uncompressed formats
- Converting AAC audio captured on an iPhone into AIFC so it can be processed by legacy Apple audio software or workflows built around the AIFF/AIFC ecosystem
- Expanding dynamic range headroom before applying mastering plugins by moving from lossy AAC to a full 16-bit PCM representation that won't add further compression artifacts during processing
- Archiving AAC audio in a well-documented, uncompressed container format for long-term storage in Apple-centric media preservation workflows
- Supplying uncompressed audio to broadcast or post-production pipelines that reject lossy input formats like AAC for compliance reasons
Frequently Asked Questions
No. Converting AAC to AIFC with PCM audio does not restore quality lost during AAC encoding. The AAC decoder produces the best possible reconstruction of the original audio, and that reconstructed signal is written losslessly into the AIFC container — but the lossy artifacts from the original AAC compression are already baked in. You gain no quality headroom beyond what the AAC file already contained; you simply stop adding further lossy degradation from any additional encoding steps.
AAC achieves its small size through lossy perceptual compression, often achieving roughly 10:1 compression ratios compared to uncompressed audio. The pcm_s16be codec used in the output AIFC file stores every audio sample as a raw 16-bit integer with no compression, so a 5 MB AAC file might expand to 50 MB or more as AIFC. This size increase is expected and is the nature of moving from a lossy compressed format to lossless uncompressed PCM.
pcm_s16be stands for signed 16-bit PCM with big-endian byte order, which is the native PCM format of the AIFF/AIFC container. It provides a 96 dB dynamic range and is the standard CD-quality audio depth, fully sufficient for most AAC source material since consumer AAC files rarely exceed the quality ceiling of 16-bit anyway. If your downstream workflow specifically requires 24-bit or 32-bit depth (for example, for mixing headroom), you can modify the FFmpeg command to use pcm_s24be or pcm_s32be instead.
Metadata support varies. Basic ID3-style tags embedded in an AAC file (such as title and artist) may be partially preserved depending on how FFmpeg maps them to AIFC's chunk-based metadata structure, but AIFC has limited native metadata support compared to modern containers. Album art is very unlikely to be carried over. If metadata preservation is critical, verify the output tags with a tool like MediaInfo or Kid3 after conversion.
For pcm_s16be, the -b:a bitrate flag has limited practical effect since PCM audio has a fixed bit rate determined by sample rate and bit depth rather than a variable encoder setting. To change the bit depth, replace pcm_s16be with another supported codec such as pcm_s24be for 24-bit or pcm_f32be for 32-bit float. To change the sample rate, add -ar 48000 (or your desired rate) before the output filename. For example: ffmpeg -i input.aac -c:a pcm_s24be -ar 48000 output.aifc
Yes. On Linux or macOS, you can use a shell loop: for f in *.aac; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.aac}.aifc"; done. On Windows Command Prompt, use: for %f in (*.aac) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc". The browser tool processes one file at a time, but the displayed FFmpeg command is designed so you can adapt it for bulk desktop processing — especially useful for large libraries that exceed the 1 GB browser limit.
Technical Notes
The AIFC format (Audio Interchange File Format Compressed) is a superset of Apple's AIFF specification and uses big-endian byte ordering, which reflects its origins on Motorola 68k-based Macintosh systems. When FFmpeg writes pcm_s16be into an AIFC container, it technically produces a valid AIFC file even though the audio is uncompressed — AIFC's 'COMM' chunk identifies the compression type as 'NONE' in this case. The source AAC codec can be either the native FFmpeg aac decoder or libfdk_aac depending on your build, but both produce equivalent decoded PCM for this purpose. One known limitation is that AIFC does not support multiple audio tracks, subtitles, chapters, or embedded cover art, so any of these present in a wrapped AAC file (such as one inside an M4A container renamed to .aac) will be silently dropped. Sample rate is preserved from the source AAC file by default. If the source AAC was encoded at a low bitrate (64k or 96k), the resulting AIFC will faithfully reproduce the limited-quality decoded audio at full PCM resolution — increasing file size substantially without any perceptual benefit, so it is worth considering whether the original AAC quality is adequate before committing to this conversion.