Convert ALAC to AIFC — Free Online Tool
Convert ALAC audio files (.m4a) to AIFC format by decoding Apple's lossless compression and re-encoding the audio as 16-bit big-endian PCM (pcm_s16be) — a format widely used in professional audio workflows on macOS and compatible with legacy Apple and broadcast tools.
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FFmpeg Command
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DownloadHow It Works
ALAC (Apple Lossless Audio Codec) stores audio in a losslessly compressed form inside an MPEG-4 container (.m4a). During this conversion, FFmpeg fully decodes the ALAC-compressed audio back to raw PCM samples, then re-encodes it as 16-bit signed big-endian PCM (pcm_s16be) and wraps it in an AIFC container. AIFC is an extension of Apple's AIFF format that supports both uncompressed and compressed audio data. Because ALAC is lossless, the decoded PCM is a bit-perfect representation of the original source — however, if your ALAC file was recorded at 24-bit depth, storing it in 16-bit pcm_s16be will reduce bit depth, which is a one-way quality reduction. The resulting AIFC file will be significantly larger than the ALAC source because pcm_s16be is uncompressed PCM.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all decoding, encoding, and container remuxing for this conversion. |
-i input.m4a
|
Specifies the input file — an MPEG-4 container (.m4a) carrying an ALAC-encoded audio stream. FFmpeg will detect the ALAC codec automatically and decode it to raw PCM for processing. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding for AIFC files. This fully decodes the ALAC compression and re-encodes the audio as raw, uncompressed samples in AIFC-native byte order. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps, but this flag has no practical effect when encoding to pcm_s16be because uncompressed PCM has a fixed bitrate determined solely by sample rate and bit depth. It is effectively ignored by FFmpeg in this context. |
output.aifc
|
Defines the output filename and instructs FFmpeg to write the result as an AIFC file. FFmpeg uses the .aifc extension to select the AIFF-C container format, which wraps the pcm_s16be audio stream. |
Common Use Cases
- Importing Apple Music lossless exports into professional macOS audio software (such as Logic Pro or older versions of Peak) that expects AIFF/AIFC rather than ALAC-encoded M4A files.
- Delivering audio assets to broadcast or post-production facilities that require uncompressed big-endian PCM wrapped in an AIFC container per legacy Apple/SGI interoperability standards.
- Archiving ALAC music collections in a universally readable PCM format so the audio can be opened without any Apple-specific codec support.
- Preparing audio for use with older hardware samplers or digital audio workstations that read AIFC but do not have ALAC decoding capability.
- Converting iTunes-purchased or Apple Music lossless tracks to AIFC for compatibility with scientific audio analysis tools that expect raw PCM in an AIFF-family container.
- Stripping the MPEG-4 container from ALAC files when a workflow requires AIFF-family containers for metadata or scripting reasons on macOS systems.
Frequently Asked Questions
If your ALAC source was originally recorded or stored at 16-bit depth, the conversion to pcm_s16be is lossless — every audio sample is preserved exactly. However, if the ALAC file is 24-bit (common with high-resolution downloads or recordings), downconverting to 16-bit pcm_s16be will reduce dynamic range and discard the lower 8 bits of each sample. To avoid this, you can change the codec to pcm_s24be in the FFmpeg command to match the original 24-bit depth.
ALAC uses lossless compression that typically reduces file size by 40–60% compared to raw PCM, so a 50 MB ALAC file might expand to 80–100 MB as uncompressed AIFC. The AIFC container in this conversion uses pcm_s16be, which stores every audio sample as raw, uncompressed 16-bit big-endian integers with no compression at all. This is expected and normal — you are trading storage efficiency for a universally readable, codec-independent audio format.
ALAC files stored in the MPEG-4 container can carry rich iTunes-style metadata (ID3-like atoms). The AIFC format has a more limited metadata structure based on AIFF chunks, and FFmpeg will attempt to map common tags during conversion, but not all metadata fields are guaranteed to survive. Embedded artwork and custom Apple-specific tags are particularly likely to be lost or ignored. If metadata preservation is critical, verify the output file's tags with a tool like MediaInfo after conversion.
The -b:a 128k flag sets a target audio bitrate, but for uncompressed PCM codecs like pcm_s16be, the bitrate is determined entirely by the sample rate and bit depth — it is not a tunable parameter the way it is for lossy codecs. FFmpeg will simply ignore the -b:a flag when encoding to pcm_s16be, so leaving it in the command is harmless but has no effect. You can safely remove it from the command if you run it locally.
Replace -c:a pcm_s16be with -c:a pcm_s24be in the command: ffmpeg -i input.m4a -c:a pcm_s24be output.aifc. This instructs FFmpeg to encode the audio as 24-bit signed big-endian PCM, which preserves the full dynamic range of high-resolution ALAC files and avoids any bit-depth reduction. The pcm_s24be codec is also a valid AIFC audio format and is commonly used in professional audio contexts.
Yes. On macOS or Linux, you can use a shell loop to process an entire folder: for f in *.m4a; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4a}.aifc"; done. On Windows Command Prompt, use: for %f in (*.m4a) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc". Each file is processed sequentially, and the output filename is derived from the input filename with the extension changed to .aifc.
Technical Notes
ALAC and AIFC are both historically tied to Apple's ecosystem, but they serve very different purposes. ALAC is a compressed lossless codec designed for efficient storage in MPEG-4 containers, while AIFC is an uncompressed (in this configuration) container format rooted in the AIFF specification originally developed by Apple and Electronic Arts in the late 1980s. The pcm_s16be codec used in this conversion is big-endian, meaning the most significant byte comes first — this is the native byte order for AIFF/AIFC and for older PowerPC-based Macs, but differs from the little-endian byte order used in WAV files on modern x86/ARM systems. The AIFC format technically supports compressed audio (including A-law and mu-law via pcm_alaw and pcm_mulaw), but the most common and compatible usage is uncompressed PCM. Note that AIFC does not support chapters, so any chapter markers embedded in your ALAC M4A file will be discarded during conversion. Similarly, multiple audio tracks and subtitle streams are not supported by AIFC, though ALAC M4A files rarely carry these. For professional audio production, the sample rate from the ALAC source is preserved automatically by FFmpeg unless you explicitly specify -ar to resample.