MyMediaTools

Convert FLAC to AIF — Free Online Tool

Convert FLAC lossless compressed audio to AIF (AIFF) format by decoding the FLAC stream and encoding it as uncompressed 16-bit big-endian PCM — the native audio format for Apple and professional Mac-based workflows. Both formats are lossless, so no audio quality is lost during conversion.

FFmpeg Command

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How It Works

FLAC stores audio as losslessly compressed data using its own internal codec, which must be fully decoded before conversion. During this process, FFmpeg decodes the FLAC bitstream into raw PCM samples, then re-encodes them as 16-bit big-endian PCM (pcm_s16be) inside an AIFF container. Because both formats are lossless, the decoded PCM samples are mathematically identical to the originals — there is no generation loss. The main change is that AIF/AIFF is entirely uncompressed, so the resulting file will be significantly larger than the FLAC source. If your FLAC file was recorded or mastered at higher bit depths (24-bit or 32-bit), the default command encodes to 16-bit, which involves bit-depth reduction; you can override this with a different codec flag to preserve higher bit depths.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg command-line tool. In this browser-based tool, the equivalent FFmpeg.wasm (WebAssembly) binary runs locally in your browser, processing your file without uploading it to any server.
-i input.flac Specifies the input file — a FLAC audio file. FFmpeg reads the FLAC container, identifies the compressed FLAC audio stream inside, and prepares to decode it into raw PCM samples for conversion.
-c:a pcm_s16be Sets the output audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding used inside AIFF files. The 'be' (big-endian) byte order is required by the AIFF specification and differentiates it from WAV's little-endian PCM. If your source FLAC is 24-bit, consider using pcm_s24be here to avoid bit-depth reduction.
output.aif Specifies the output filename and container format. The .aif extension tells FFmpeg to wrap the pcm_s16be audio stream in an AIFF container — Apple's Audio Interchange File Format — producing a lossless, uncompressed audio file compatible with Logic Pro, GarageBand, Final Cut Pro, and other professional Mac audio tools.

Common Use Cases

  • Importing audio into Logic Pro, GarageBand, or Final Cut Pro, which historically prefer or natively handle AIFF over FLAC for editing and scoring projects.
  • Delivering audio assets to a Mac-based post-production studio or mastering engineer whose DAW workflow is built around AIFF as the interchange format.
  • Archiving or distributing audio to older Apple hardware or software (such as early iTunes versions) that can play AIFF natively but may not support FLAC playback.
  • Preparing audio samples or loops for use in Ableton Live, Logic Pro, or Native Instruments samplers that import AIFF more reliably than FLAC in certain plugin contexts.
  • Converting a FLAC music library to AIFF for playback on high-fidelity audiophile hardware or network streamers that support AIFF/AIFF-C but lack a FLAC decoder.
  • Creating an uncompressed AIFF master from a FLAC source as an intermediate step before further audio processing in professional audio tools that require raw PCM input.

Frequently Asked Questions

No — both FLAC and AIF/AIFF are lossless formats, so the conversion does not introduce any audible degradation or lossy compression artifacts. The FLAC decoder reconstructs the exact PCM samples that were originally compressed, and those samples are written directly into the uncompressed AIFF container. The only potential quality consideration is bit depth: if your FLAC file is 24-bit and you use the default pcm_s16be codec, the output is reduced to 16-bit. To preserve 24-bit depth, use '-c:a pcm_s24be' in the FFmpeg command instead.
FLAC uses lossless compression to reduce file size — typically achieving 40–60% size reduction compared to raw PCM. AIF/AIFF, by contrast, stores audio as completely uncompressed PCM data with no compression applied. When you decode the FLAC and write it as uncompressed AIFF, the file expands back to its full uncompressed size. For example, a 30MB FLAC file might expand to 70–100MB as an AIFF. This is expected and does not indicate any problem with the conversion.
Partially. FLAC uses Vorbis comment-style tags, while AIFF uses a different metadata chunk structure (ID3 tags embedded in a MARK or NAME chunk). FFmpeg will attempt to map common tags such as title, artist, and album into the AIFF container, but less common or format-specific tags (such as FLAC's REPLAYGAIN fields or cue sheet data) are unlikely to be preserved. If precise metadata portability matters, verify the output tags with a tool like MediaInfo or a tag editor after conversion.
Replace '-c:a pcm_s16be' with '-c:a pcm_s24be' in the command: 'ffmpeg -i input.flac -c:a pcm_s24be output.aif'. This encodes the audio as 24-bit big-endian PCM, which is the correct choice if your FLAC source was recorded or mastered at 24-bit resolution. Using pcm_s24be avoids any bit-depth reduction and produces a higher-resolution AIFF suitable for professional audio work. Other supported bit-depth options include pcm_s32be and pcm_f32be for 32-bit output.
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.flac; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.flac}.aif"; done'. On Windows Command Prompt, use: 'for %f in (*.flac) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif"'. This iterates over every FLAC file in the current directory and produces a corresponding AIF file with the same base filename. The browser-based tool on this page processes one file at a time, so the FFmpeg command approach is particularly useful for bulk library conversions.
AIF and AIFF refer to the same Audio Interchange File Format developed by Apple — the difference is purely in the file extension (.aif vs .aiff), and both extensions reference identical file structures. While AIFF was designed by Apple and is most commonly associated with Mac workflows, it is not Mac-exclusive: Windows applications including Audacity, Adobe Audition, and VLC all support AIFF playback and editing natively. The uncompressed PCM data inside AIFF is also one of the most universally readable audio formats across professional audio software.

Technical Notes

The conversion from FLAC to AIF involves a full decode-and-re-encode cycle rather than a simple remux, because FLAC's compressed bitstream is fundamentally incompatible with the raw PCM data that AIFF stores. The default output codec is pcm_s16be — 16-bit signed, big-endian PCM — which matches the original AIFF specification and is the most compatible option across Apple hardware and software. Big-endian byte ordering is a distinguishing characteristic of AIFF (compared to WAV, which uses little-endian PCM); FFmpeg handles this byte-order requirement automatically when the output format is .aif or .aiff. If your FLAC source has a sample rate other than 44.1kHz or 48kHz (for example, 88.2kHz or 96kHz hi-res audio), FFmpeg preserves the original sample rate in the AIFF output without resampling unless you explicitly request it. FLAC's ReplayGain tags and embedded cue sheets are not natively representable in AIFF's metadata structure and will be dropped during conversion. Channel layouts (stereo, mono, multichannel) are preserved. AIFF-C, the compressed variant of AIFF, is a distinct format and is not produced by this command — the output is standard uncompressed AIFF.

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