Convert AIFF to FLAC — Free Online Tool
Convert AIFF files to FLAC while preserving every bit of your original audio. This tool re-encodes Apple's uncompressed PCM audio (pcm_s16be, pcm_s24be, etc.) into FLAC's lossless compression, shrinking file sizes by 40–60% with zero audible or mathematical quality loss.
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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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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
AIFF stores raw, uncompressed PCM audio data in big-endian byte order — the same linear pulse-code modulation used in professional studios, just written directly to disk with no compression. FLAC conversion takes that raw PCM stream and applies a lossless predictor-based compression algorithm (similar to ZIP but designed for audio waveforms), encoding residuals between predicted and actual sample values. The output FLAC file contains mathematically identical audio samples to the source AIFF — bit-for-bit the same waveform — but stored far more efficiently. Because both formats are lossless, no frequency information, dynamic range, or transient detail is sacrificed. The FLAC encoder also embeds a MD5 checksum of the original audio, so players can verify the decoded audio matches the source exactly.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing engine, which handles all reading, decoding, encoding, and writing operations for this AIFF-to-FLAC conversion. |
-i input.aiff
|
Specifies the input AIFF file. FFmpeg reads the file's IFF container, detects the PCM audio codec (such as pcm_s16be or pcm_s24be), and decodes the raw audio samples into memory for re-encoding. |
-c:a flac
|
Instructs FFmpeg to encode the audio stream using the FLAC lossless codec, replacing the uncompressed big-endian PCM data from the AIFF with FLAC's losslessly compressed representation of the same audio samples. |
-compression_level 5
|
Sets the FLAC encoder's compression effort to level 5 (out of 0–8), which is the standard default. This controls encoding speed and output file size only — all compression levels produce decoded audio that is bit-for-bit identical to the source AIFF PCM data. |
output.flac
|
Defines the output filename with the .flac extension, telling FFmpeg to write the encoded audio into a FLAC container. The resulting file will be a standards-compliant FLAC file playable on virtually any audio software or device that supports lossless audio. |
Common Use Cases
- Archiving a macOS GarageBand or Logic Pro session's exported AIFF masters into a more storage-efficient format without degrading the 24-bit or 32-bit source audio.
- Uploading high-resolution audio to music distribution platforms or streaming services that accept FLAC but not AIFF.
- Moving a large AIFF sample library (drum hits, orchestral samples, etc.) to a portable drive or NAS where space is limited, without compromising the samples' original quality.
- Sharing lossless audio recordings with collaborators on Windows or Linux, where FLAC has broader native support than Apple's AIFF format.
- Preparing 24-bit AIFF masters for submission to a mastering engineer who requires FLAC with embedded metadata tags for artist, album, and track information.
- Reducing the storage footprint of a collection of lossless AIFF recordings from a field recorder or digital audio workstation before long-term cold storage.
Frequently Asked Questions
No — both AIFF and FLAC are lossless formats, so the conversion preserves the audio data perfectly. FLAC's compression is mathematically reversible: every sample value decoded from the FLAC file will be identical to the original PCM samples in the AIFF. FLAC even embeds an MD5 checksum to verify this integrity. You can safely convert 16-bit, 24-bit, or 32-bit AIFF files to FLAC without any degradation.
FLAC typically reduces file size by 40–60% compared to uncompressed AIFF. The exact ratio depends on the audio content — highly dynamic material like classical music or sparse acoustic recordings compresses more efficiently than dense, maximally-loud electronic music. A 100 MB AIFF file might become 45–60 MB as FLAC. The compression level setting (0–8) controls the trade-off between encoding speed and compression ratio, though all levels produce identical audio output.
The -compression_level flag controls how hard the FLAC encoder works to shrink the file, on a scale from 0 (fastest, largest) to 8 (slowest, smallest). Level 5 is the widely accepted default that balances encoding speed with good compression. Crucially, all levels from 0 to 8 produce bit-for-bit identical decoded audio — changing the level only affects file size and encoding time, never audio quality. For most users, level 5 is ideal; only bump it to 7 or 8 if minimizing file size is critical and you don't mind slower encoding.
AIFF stores metadata in its ID3 or MARK/NAME chunks, while FLAC uses Vorbis comment tags. FFmpeg will attempt to map common fields like title, artist, album, and track number during conversion, but some AIFF-specific metadata fields (such as instrument loop points or MIDI data stored in AIFF chunks) have no FLAC equivalent and will be discarded. Standard text metadata is generally preserved correctly.
Yes — on Linux or macOS, you can loop over all AIFF files in a directory with a shell one-liner: for f in *.aiff; do ffmpeg -i "$f" -c:a flac -compression_level 5 "${f%.aiff}.flac"; done. On Windows Command Prompt, use: for %f in (*.aiff) do ffmpeg -i "%f" -c:a flac -compression_level 5 "%~nf.flac". This applies the same settings — FLAC codec, compression level 5 — to every file in the folder.
FLAC natively supports bit depths up to 32 bits per sample and sample rates up to 655,350 Hz, so it can fully represent 16-bit, 24-bit, and 32-bit AIFF content without truncation. However, FLAC's integer-only encoding means pcm_f32be or pcm_f64be floating-point AIFF files will be converted to integer PCM during encoding — FFmpeg handles this conversion automatically, but be aware that 32-bit float AIFF and 32-bit integer FLAC represent audio slightly differently in terms of dynamic range headroom.
Technical Notes
AIFF encodes audio as big-endian PCM, which is why its common codecs (pcm_s16be, pcm_s24be, pcm_s32be) carry the 'be' suffix — this is the native byte order of classic Motorola and PowerPC processors that Apple historically used. FLAC, by contrast, is always integer-based and platform-agnostic in its encoding. When FFmpeg converts a floating-point AIFF (pcm_f32be or pcm_f64be) to FLAC, it must first convert the floating-point samples to integers, which is handled transparently but worth knowing if your source material was recorded or processed in 32-bit float. FLAC's compression level 5 default produces files that are typically 40–60% smaller than the source AIFF, with encoding taking only a few seconds per minute of audio on modern hardware. One notable limitation: FLAC does not support Apple's AIFF loop point or instrument chunk metadata, which is relevant for sample library creators. FLAC does, however, support ReplayGain tags and cue sheets, making it well-suited for archival and audiophile use cases. Both formats support up to 24-bit resolution natively in their most common use cases, ensuring professional-grade audio fidelity is maintained end-to-end.