Convert AIFC to FLAC — Free Online Tool
Convert AIFC files to FLAC with true lossless quality — whether your AIFC contains uncompressed PCM or compressed audio data, this tool re-encodes it into the universally supported FLAC format using the FLAC codec at compression level 5, preserving every sample with no audible degradation.
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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
AIFC is Apple's compressed extension of AIFF, capable of storing audio in several PCM variants (16-bit, 24-bit, 32-bit, floating-point) as well as lossy formats like A-law and μ-law (mu-law). During conversion, FFmpeg decodes the AIFC audio stream — unpacking whatever codec was used inside the container — and re-encodes it into FLAC, a lossless compressed format. If the source AIFC contains uncompressed PCM (the most common case), the conversion is fully lossless end-to-end: FLAC captures an exact mathematical representation of the original waveform while reducing file size by 40–60%. If the source uses a lossy codec like A-law or μ-law, the output FLAC will be a lossless capture of that already-degraded signal — no further quality is lost, but the original lossy degradation is preserved. FLAC compression level 5 (the default) balances encode speed against file size without affecting audio fidelity.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which is the underlying engine handling all decoding, sample format conversion, and re-encoding in this AIFC-to-FLAC pipeline. |
-i input.aifc
|
Specifies the input AIFC file. FFmpeg will detect the AIFC container and select the appropriate decoder for whatever audio codec is stored inside — whether that's uncompressed PCM (pcm_s16be, pcm_s24be) or a compressed variant like A-law or μ-law. |
-c:a flac
|
Instructs FFmpeg to encode the decoded audio stream using the FLAC codec. This produces lossless compressed audio in FLAC format, replacing whatever codec (PCM, A-law, μ-law, etc.) was present in the original AIFC container. |
-compression_level 5
|
Sets FLAC's compression level to 5, the standard default, which balances encode speed and output file size. This has absolutely no effect on audio quality — all FLAC compression levels are lossless — but higher values produce slightly smaller files at the cost of longer encode times. |
output.flac
|
Defines the output filename and its .flac extension, which tells FFmpeg to write the result into a FLAC container. The encoded FLAC audio stream is muxed into this file, ready for use in any FLAC-compatible player, DAW, or archival system. |
Common Use Cases
- Archiving Apple Logic Pro or older Mac-era audio recordings stored as AIFC files into the open, widely supported FLAC format for long-term preservation
- Preparing AIFC audio samples or sound design assets for use in software that reads FLAC but not AIFC, such as many Linux-based DAWs and media players
- Converting AIFC files exported from legacy ProTools or Final Cut Pro sessions into FLAC for distribution on audiophile music platforms like Bandcamp or Qobuz
- Migrating a library of 24-bit AIFC recordings from macOS to a cross-platform workflow where FLAC is the standard lossless container
- Reducing the storage footprint of large uncompressed 32-bit AIFC audio files by re-encoding to FLAC without sacrificing any audio data
- Preparing AIFC dialogue or foley recordings for handoff to a post-production team using Windows or Linux tools that have broader FLAC support than AIFC
Frequently Asked Questions
If your AIFC file contains uncompressed PCM audio (which is the default and most common case for AIFC), the conversion to FLAC is entirely lossless — not a single audio sample changes. FLAC uses lossless compression, meaning the decoded output is bit-for-bit identical to the input waveform. The only scenario where quality is already compromised is if your AIFC file was encoded with a lossy codec like A-law or μ-law — in that case, FLAC preserves the signal exactly as it exists, but cannot recover quality lost during the original lossy encoding.
Uncompressed AIFC files store raw PCM audio data with no size reduction at all, while FLAC applies lossless compression algorithms (similar in concept to ZIP, but optimized for audio waveforms) to reduce file size by roughly 40–60% without altering the audio data. This means a 100 MB uncompressed AIFC file might become a 45–60 MB FLAC file that decodes to the exact same audio. The compression level 5 used by this tool is the standard middle-ground that achieves good size reduction without unusually long encode times.
FLAC natively supports bit depths of 4 to 24 bits, so 16-bit and 24-bit AIFC files convert without any bit-depth reduction. However, if your AIFC file uses 32-bit integer or 64-bit floating-point PCM (pcm_s32be or pcm_f64be), FLAC cannot represent those at full depth — FFmpeg will typically downscale these to 24-bit during the encode. For the vast majority of professional audio work, 24-bit is the practical ceiling anyway, so this is rarely a concern in practice.
AIFC stores metadata in a limited set of AIFF-style chunk fields, and FLAC uses a flexible Vorbis comment tagging system. FFmpeg will attempt to map compatible AIFC metadata tags to their FLAC equivalents during conversion, but AIFC's metadata support is considerably more limited than FLAC's, so you may need to re-tag your FLAC files afterward using a tool like MusicBrainz Picard or beets. FLAC's Vorbis comments support arbitrary tags, replay gain values, and embedded artwork, giving you more flexibility post-conversion.
The compression level is controlled by the -compression_level flag in the command. You can set it to any integer from 0 (fastest encode, largest file) to 8 (slowest encode, smallest file). For example, to use maximum compression, change the command to: ffmpeg -i input.aifc -c:a flac -compression_level 8 output.flac. Crucially, this setting only affects encode speed and output file size — all compression levels from 0 to 8 produce identical audio quality since FLAC is always lossless.
The single-file command shown is a starting point, but FFmpeg itself doesn't have a built-in batch mode. On macOS or Linux, you can wrap it in a shell loop: for f in *.aifc; do ffmpeg -i "$f" -c:a flac -compression_level 5 "${f%.aifc}.flac"; done. On Windows Command Prompt, use: for %f in (*.aifc) do ffmpeg -i "%f" -c:a flac -compression_level 5 "%~nf.flac". The browser tool on this page processes one file at a time, so the FFmpeg command is the best path for large batch jobs.
Technical Notes
AIFC (Audio Interchange File Format Compressed) is a big-endian format, meaning its PCM variants (pcm_s16be, pcm_s24be, pcm_s32be) store bytes in big-endian byte order, which is native to older PowerPC Macs but requires byte-swapping on modern little-endian processors. FFmpeg handles this transparently during decode. FLAC, by contrast, defines its own internal binary structure independent of host endianness. One important limitation to be aware of: FLAC's lossless compression is defined only for integer PCM samples up to 24 bits. If your AIFC source uses floating-point PCM (pcm_f32be or pcm_f64be), FFmpeg will convert the floating-point samples to 24-bit integer representation during encoding, which can introduce a negligible quantization difference at the noise floor — well below audibility at 24 bits. AIFC files using A-law or μ-law compression are 8-bit lossy codecs used historically in telephony; converting these to FLAC will produce a lossless copy of already low-fidelity 8-bit audio. FLAC's seeking, cue sheet, and replay gain features make it significantly more functional than AIFC for music library management. Metadata portability between AIFC and FLAC is partial — common fields like title and artist transfer, but AIFC-specific markers and instrument chunks have no FLAC equivalent and will be discarded.