Convert FLAC to AIFC — Free Online Tool
Convert FLAC lossless audio to AIFC format using PCM big-endian encoding directly in your browser. This tool decodes your FLAC file and re-encodes it as 16-bit signed big-endian PCM (pcm_s16be) inside an AIFC container — the standard uncompressed audio format used in professional Apple and broadcast workflows.
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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
FLAC stores audio as losslessly compressed data that must be fully decoded before any container conversion. During this conversion, FFmpeg decompresses the FLAC bitstream back to raw PCM audio samples, then re-encodes those samples as 16-bit signed big-endian PCM (pcm_s16be) and wraps them in an AIFC container. AIFC is an extension of Apple's AIFF format that supports both uncompressed PCM and compressed codecs — here we use uncompressed PCM, so there is no generation loss. The resulting file is larger than the source FLAC because PCM carries no compression, but the audio data itself is mathematically identical in quality. Metadata tags from the FLAC file may not carry over to AIFC, as the two formats use different metadata schemes (Vorbis comments vs. AIFF markers and annotations).
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles decoding the FLAC input and encoding the AIFC output. In the browser version, this runs via FFmpeg.wasm compiled to WebAssembly. |
-i input.flac
|
Specifies the input file — in this case a FLAC file containing losslessly compressed audio. FFmpeg will fully decompress the FLAC bitstream to raw PCM samples before re-encoding. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM, the standard uncompressed audio encoding used inside AIFC and AIFF containers. This produces bit-perfect audio from a 16-bit FLAC source with no compression or quality reduction. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps, but this parameter has no practical effect on pcm_s16be since uncompressed PCM has a fixed bitrate determined entirely by sample rate, bit depth, and channel count — not a configurable target bitrate. |
output.aifc
|
Defines the output filename with the .aifc extension, which tells FFmpeg to mux the encoded PCM audio into an AIFC container — the Apple extended audio interchange format that supports both uncompressed and compressed PCM variants. |
Common Use Cases
- Preparing lossless audio for import into Logic Pro or older Pro Tools sessions that expect AIFF/AIFC format rather than FLAC
- Delivering master audio files to broadcast or post-production facilities that require big-endian PCM in an AIFC container for compatibility with their ingest systems
- Archiving FLAC music collection tracks into AIFC for use with legacy Apple hardware or software that predates native FLAC support
- Converting FLAC soundtrack stems to AIFC so they can be imported into Final Cut Pro or older versions of GarageBand without codec issues
- Supplying uncompressed AIFC audio to mastering engineers or duplication houses whose workflows are built around the AIFF/AIFC standard rather than FLAC
Frequently Asked Questions
No — provided your FLAC source is already 16-bit audio, the conversion to pcm_s16be AIFC is completely lossless. FLAC is a lossless compressed format, and pcm_s16be is uncompressed PCM, so decoding FLAC and storing the result as raw 16-bit big-endian PCM is a bit-perfect round trip. If your FLAC source was recorded at 24-bit depth, however, the default pcm_s16be codec will truncate the extra 8 bits — in that case you should use pcm_s24be instead to preserve full depth.
FLAC achieves its small file size through lossless compression — it removes mathematical redundancy in the audio signal and restores it perfectly on playback. AIFC with pcm_s16be stores raw, uncompressed PCM samples with no compression at all. A typical FLAC file compresses audio to 50–70% of its raw size, so converting back to uncompressed PCM will roughly double or triple the file size. The audio content is identical; only the storage efficiency changes.
AIFF (Audio Interchange File Format) only supports uncompressed PCM audio, while AIFC (AIFF-Compressed) is an extension that supports both uncompressed PCM and compressed codecs like A-law and mu-law. This tool produces an AIFC file using pcm_s16be — uncompressed 16-bit big-endian PCM — so functionally the audio is identical to a standard AIFF, but the container is technically AIFC. Most professional audio software that reads AIFF will also read AIFC without any issues.
Likely not fully. FLAC uses Vorbis comment blocks for metadata, while AIFC uses a different chunk-based scheme for annotations and markers. FFmpeg will attempt to map common tags like title, artist, and album, but the translation is imperfect and some tags may be dropped. If preserving metadata is critical, check the output in an audio tag editor like Kid3 or iTunes after conversion and re-enter any missing fields manually.
Replace 'pcm_s16be' in the '-c:a' flag with another codec supported by AIFC. For 24-bit depth use 'pcm_s24be', for 32-bit float use 'pcm_f32be', and for telephony-grade compressed audio use 'pcm_alaw' or 'pcm_mulaw'. For example: 'ffmpeg -i input.flac -c:a pcm_s24be output.aifc' produces a 24-bit big-endian AIFC file, which is better suited for preserving hi-res FLAC sources recorded above 16-bit depth.
Yes — on the command line you can use a shell loop to process multiple files. On Linux or macOS, run: 'for f in *.flac; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.flac}.aifc"; done'. On Windows PowerShell, use: 'Get-ChildItem *.flac | ForEach-Object { ffmpeg -i $_.FullName -c:a pcm_s16be -b:a 128k ($_.BaseName + ".aifc") }'. The browser-based tool processes one file at a time, so the FFmpeg command is especially useful when you have a large batch of FLAC files to convert.
Technical Notes
AIFC with pcm_s16be uses big-endian byte ordering, which is native to Motorola and SPARC architectures and was the standard on classic Mac hardware. Modern x86 systems are little-endian, but FFmpeg handles the byte-order conversion transparently. The '-b:a 128k' flag in the command has no practical effect on uncompressed PCM codecs like pcm_s16be — bitrate is implicit in the sample rate and bit depth (e.g., 44100 Hz × 16 bits × 2 channels = 1,411 kbps). It is included in the resolved command template but does not alter the output. FLAC supports sample rates up to 655,350 Hz and bit depths of 4–32 bits; AIFC with pcm_s16be is limited to 16-bit depth, so sources recorded at 24-bit or 32-bit will be downsampled in bit depth unless you override the codec to pcm_s24be or pcm_s32be. FLAC's ReplayGain and cue sheet metadata have no direct equivalent in AIFC and will be silently discarded during conversion. AIFC files produced by this tool are compatible with Logic Pro, GarageBand, Final Cut Pro, and most professional DAWs and broadcast ingest systems that support the AIFF/AIFC family.