Convert AIF to FLAC — Free Online Tool
Convert AIF files to FLAC with zero quality loss — both formats are lossless, so your original PCM audio data is preserved bit-perfectly while FLAC's compression algorithm shrinks the file size by 40–60%. The output uses the FLAC codec with compression level 5, balancing encode speed and file size reduction.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AIF file here
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Free — no uploads, no signups. Your files never leave your browser.
Settings
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
AIF stores audio as raw uncompressed PCM data (typically 16-bit, 24-bit, or 32-bit big-endian samples), which means every sample value is written directly to disk with no compression whatsoever. During conversion, FFmpeg reads those raw PCM samples from the AIF container and passes them through the FLAC encoder, which applies a lossless prediction and Rice coding algorithm to reduce file size without altering a single sample value. The resulting FLAC file contains mathematically identical audio to the source AIF — decoding it will reproduce the exact original PCM waveform. The AIF container metadata (artist, title, etc.) is mapped to FLAC's Vorbis comment tag structure where possible, though AIF-specific chunk data that has no FLAC equivalent may be dropped.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. This runs entirely in your browser via FFmpeg.wasm (WebAssembly) — no files leave your device. |
-i input.aif
|
Specifies the input AIF file. FFmpeg detects the AIFF container and identifies the PCM audio stream (which may be 16-bit, 24-bit, or 32-bit big-endian depending on the source file) before passing it to the encoder. |
-c:a flac
|
Tells FFmpeg to encode the audio stream using the FLAC (Free Lossless Audio Codec) encoder. Since AIF contains only audio and no video stream, this is the only codec instruction needed — it replaces the raw uncompressed PCM from the AIF with losslessly compressed FLAC data. |
-compression_level 5
|
Sets the FLAC encoder's compression effort to level 5, which is the FLAC reference encoder's default. It controls the trade-off between encoding speed and output file size — level 5 typically reduces the AIF file size by 40–60% and encodes quickly. Audio quality is not affected by this setting. |
output.flac
|
Defines the output filename and container. The .flac extension tells FFmpeg to write a standard FLAC file, which wraps the encoded audio stream along with any metadata translated from the source AIF's chunk structure. |
Common Use Cases
- Archiving a large Apple Logic Pro session's exported AIF masters into FLAC to reclaim disk space without sacrificing the original 24-bit resolution for future remixing or mastering.
- Preparing high-resolution audio files for distribution on platforms like Bandcamp or Qobuz that accept FLAC but not AIF, while keeping full lossless fidelity for audiophile listeners.
- Converting a library of AIF samples from an older Mac-era sample pack into FLAC so they can be used in cross-platform digital audio workstations and sample players on Windows or Linux.
- Replacing AIF files on a media server or NAS (such as a Plex or Roon library) with FLAC equivalents, since FLAC has broader device and software support while being natively seekable and taggable.
- Reducing storage overhead when backing up a professional field recording archive — converting uncompressed 32-bit float AIF recordings from a Sound Devices recorder to FLAC to cut backup drive usage roughly in half.
- Sharing high-quality audio with collaborators who use non-Apple tools, since FLAC is universally supported while AIF can require additional codecs or plugins on Windows and Linux systems.
Frequently Asked Questions
No — this is a lossless-to-lossless conversion, so the audio quality is identical before and after. FLAC uses a mathematically reversible compression algorithm, meaning every sample from the original AIF PCM stream is reconstructed exactly when the FLAC file is decoded. You can verify this by decoding both files back to raw PCM and doing a bit-for-bit comparison — they will match perfectly.
Typically 40–60% smaller, though the exact reduction depends heavily on the audio content. Highly dynamic material like orchestral recordings or sparse acoustic tracks compresses more efficiently than dense electronic music or heavily limited masters. A 100 MB 24-bit stereo AIF file commonly results in a 45–60 MB FLAC file at compression level 5. The compression level setting affects only encode time and file size — not audio quality.
Yes — AIF supports 16-bit (pcm_s16be), 24-bit (pcm_s24be), 32-bit integer (pcm_s32be), and 32/64-bit float PCM. FLAC natively supports up to 24-bit integer audio. If your source AIF contains 32-bit integer or 32/64-bit float samples, FFmpeg will convert them to 24-bit during encoding, which is the maximum FLAC supports. For the vast majority of professional and consumer audio (which is 16 or 24-bit), bit depth is preserved exactly.
The -compression_level flag controls the FLAC encoder's effort in finding redundant patterns in the audio data — higher values mean smaller files but slower encoding, while lower values encode faster with slightly larger output. Level 5 is the FLAC reference encoder's default and strikes a good balance. Changing it to 8 (maximum) might shave another 2–5% off the file size but takes significantly longer, while level 0 encodes almost instantly at the cost of a somewhat larger file. Audio quality is identical at every compression level.
Partially. AIF stores metadata in chunks (NAME, AUTH, ANNO, etc.) that follow the IFF/AIFF specification, while FLAC uses Vorbis comment tags (TITLE, ARTIST, ALBUM, and so on). FFmpeg maps the most common AIF metadata fields to their FLAC equivalents during conversion. However, AIF-specific or non-standard chunk data that has no direct FLAC equivalent may be silently dropped, so it is worth verifying tags in a tool like Kid3 or MusicBrainz Picard after conversion if metadata accuracy is critical.
On Linux or macOS, you can run a shell loop: `for f in *.aif; do ffmpeg -i "$f" -c:a flac -compression_level 5 "${f%.aif}.flac"; done`. On Windows Command Prompt, use: `for %f in (*.aif) do ffmpeg -i "%f" -c:a flac -compression_level 5 "%~nf.flac"`. This applies the same codec and compression settings to every AIF file in the current directory, outputting a matching FLAC file for each one.
Technical Notes
AIF (Audio Interchange File Format) is based on the IFF chunk structure and stores PCM audio in big-endian byte order — hence codec names like pcm_s16be (signed 16-bit big-endian). This big-endian PCM is natively decoded by FFmpeg and fed directly into the FLAC encoder without any intermediate format conversion for 16 and 24-bit sources. One important limitation: FLAC's maximum supported bit depth is 24 bits, so 32-bit integer or floating-point AIF files (common from certain professional recorders and DAWs) will be downconverted to 24-bit during encoding — still very high resolution, but technically not a bit-perfect round-trip for those specific source files. FLAC adds several capabilities that AIF lacks, including robust Vorbis comment tagging, native support for cue sheets, ReplayGain metadata fields, and efficient seeking in large files. The FLAC format is also an open standard with no licensing restrictions, giving it broader support across operating systems, media players, and streaming services compared to AIF, which remains primarily an Apple ecosystem format.