MyMediaTools

Convert AIFC to ALAC — Free Online Tool

Convert AIFC audio files to ALAC (Apple Lossless Audio Codec) stored in an M4A container, preserving every bit of your original audio without any quality loss. This conversion is ideal for bringing compressed or PCM-encoded AIFC files into Apple's native lossless ecosystem — fully compatible with iTunes, Apple Music, and iOS devices.

FFmpeg Command

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How It Works

AIFC is a flexible container that can hold several audio codecs, including uncompressed PCM variants (pcm_s16be, pcm_s24be, pcm_s32be) as well as compressed formats like G.711 A-law and mu-law. During this conversion, FFmpeg decodes the AIFC audio stream — whatever codec it uses internally — to raw PCM, then re-encodes it using Apple's ALAC codec and wraps the result in an MPEG-4 (.m4a) container. Because ALAC is itself lossless, any AIFC file that contains lossless PCM audio (the common case) undergoes a fully lossless round-trip: you lose no audio information whatsoever. If the source AIFC contains a lossy codec such as pcm_alaw or pcm_mulaw, the output ALAC will faithfully encode those already-degraded samples without introducing further quality loss. The MPEG-4 container also provides better metadata tag support than AIFC, and ALAC files play natively across the entire Apple ecosystem without any additional software.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg multimedia processing engine, which handles decoding the AIFC input, converting the audio stream through its codec pipeline, and writing the ALAC-encoded M4A output.
-i input.aifc Specifies the input AIFC file. FFmpeg will detect the AIFC container and identify the embedded audio codec (which may be any of the supported variants such as pcm_s16be, pcm_s24be, pcm_alaw, etc.) before decoding it to raw PCM for re-encoding.
-c:a alac Instructs FFmpeg to encode the audio stream using Apple's Lossless Audio Codec (ALAC), producing perfectly lossless compressed audio suitable for playback in iTunes, Apple Music, and all Apple devices. This flag appears twice in the generated command; the second instance supersedes the first but both produce the same result.
-c:a alac A duplicate of the preceding audio codec flag. FFmpeg applies the last matching flag value, so this effectively confirms ALAC as the encoder. It is safe to omit the first occurrence when running the command manually without changing the output.
output.m4a Defines the output filename with an .m4a extension, which signals FFmpeg to wrap the ALAC-encoded audio in an MPEG-4 container — the only container format in which ALAC is standardly stored and recognized by Apple software and devices.

Common Use Cases

  • Archiving legacy Mac or ProTools session audio exported as AIFC into a modern Apple-native lossless format that syncs with Apple Music and iCloud.
  • Preparing high-resolution 24-bit or 32-bit AIFC masters for distribution through iTunes or Apple's digital supply chain, which requires ALAC in an M4A container.
  • Converting AIFC files recorded by older Apple hardware or software (e.g., classic macOS sound tools) into a format playable on current iPhones, iPads, and HomePods without transcoding.
  • Migrating a professional audio library stored as big-endian PCM AIFC files into ALAC to reduce file size while retaining lossless fidelity and gaining richer ID3-style metadata tags.
  • Replacing AIFC audio assets in a multimedia project with ALAC equivalents that are natively supported by Final Cut Pro and GarageBand without importing workarounds.
  • Converting telephony-grade AIFC files encoded with A-law or mu-law into ALAC for long-term lossless archiving of the exact audio data those files contain.

Frequently Asked Questions

If your AIFC file contains uncompressed PCM audio (pcm_s16be, pcm_s24be, pcm_s32be, pcm_f32be, or pcm_f64be), the conversion to ALAC is completely lossless — every sample is preserved bit-for-bit. ALAC is a lossless compression codec, so it reduces file size through compression without discarding any audio data. The only exception is if your source AIFC uses a lossy codec like pcm_alaw or pcm_mulaw; in that case the ALAC output will encode those samples exactly as they are, introducing no additional degradation beyond what was already present.
ALAC is inherently tied to the MPEG-4 container format, so files using the ALAC codec are always stored with an .m4a extension. Apple designed ALAC to live inside MPEG-4, which is why the format integrates so seamlessly with iTunes, Apple Music, and iOS. M4A containers also support richer metadata fields (artist, album art, genre, year, lyrics) than the older AIFC format, which is an added benefit of this conversion.
ALAC supports bit depths of 16, 20, 24, and 32 bits per sample, which covers the most common AIFC PCM variants (pcm_s16be and pcm_s24be especially). However, AIFC can also store 32-bit and 64-bit floating-point audio (pcm_f32be, pcm_f64be), which ALAC does not natively support in floating-point form. FFmpeg will convert floating-point samples to the nearest supported integer bit depth during encoding, so extremely wide dynamic range captured in float-format AIFC files may be subtly altered at the encoding stage.
The command shown is 'ffmpeg -i input.aifc -c:a alac -c:a alac output.m4a', which specifies the ALAC audio codec flag twice. In practice, FFmpeg simply applies the last value it encounters for a given stream specifier, so the duplicate is harmless — the result is identical to using '-c:a alac' once. You can safely remove the repeated flag when running the command locally: 'ffmpeg -i input.aifc -c:a alac output.m4a' produces the exact same output.
Yes. On macOS or Linux you can use a shell loop: 'for f in *.aifc; do ffmpeg -i "$f" -c:a alac "${f%.aifc}.m4a"; done'. On Windows Command Prompt the equivalent is 'for %f in (*.aifc) do ffmpeg -i "%f" -c:a alac "%~nf.m4a"'. This is particularly useful for large audio libraries where the browser-based tool's 1 GB per-file limit would otherwise be a constraint.
FFmpeg will attempt to carry over any metadata tags embedded in the AIFC file (such as title, artist, or comment fields) into the M4A container's atom-based tag structure. However, AIFC stores metadata in a more limited way than MPEG-4, so not every custom or proprietary field may map across cleanly. Standard fields like name, author, and copyright usually transfer correctly, while non-standard AIFC chunks are likely to be dropped.

Technical Notes

AIFC uses big-endian byte ordering for its PCM variants (hence the 'be' suffix in codec names like pcm_s16be), which reflects its origin as an Apple/Motorola-architecture format. ALAC, while also an Apple format, is stored in a little-endian-friendly MPEG-4 container and handles its own internal byte ordering transparently. FFmpeg manages this endian conversion automatically during decoding, so no manual intervention is required. One important limitation to note is that ALAC does not support floating-point sample formats; if your AIFC source uses pcm_f32be or pcm_f64be, FFmpeg will convert the samples to 32-bit integer PCM before ALAC encoding, which could clip samples that exceed the integer range (though well-recorded professional audio rarely exercises this edge case). ALAC achieves typical compression ratios of 40–60% relative to raw PCM, so a 24-bit AIFC file will generally shrink noticeably while remaining bit-for-bit lossless. The M4A container supports chapter markers, which AIFC does not, so if you add chapter metadata after conversion it will be preserved in the output file.

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