Convert FLAC to AIFF — Free Online Tool
Convert FLAC lossless audio to AIFF format using PCM 16-bit big-endian encoding — Apple's native uncompressed audio standard. This tool runs entirely in your browser via FFmpeg.wasm, decoding the FLAC stream and writing raw PCM audio into an AIFF container with no quality loss whatsoever.
to
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 PCM data, meaning the actual audio samples are mathematically identical to the original recording — the compression only saves disk space. During this conversion, FFmpeg fully decodes the FLAC stream back to raw PCM audio samples and then repackages them into an AIFF container using the pcm_s16be codec (signed 16-bit big-endian PCM), which is the standard uncompressed format Apple's AIFF uses. No audio information is lost at any stage — the waveform is bit-for-bit equivalent to what was stored in the FLAC file. The trade-off is that AIFF is uncompressed, so the output file will be significantly larger than the FLAC source (typically 2–4× larger depending on the original audio content and FLAC compression level).
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In this browser-based tool, the same binary runs locally via WebAssembly (FFmpeg.wasm), so the command shown is identical to what you would run in a terminal on your own machine. |
-i input.flac
|
Specifies the input file — a FLAC audio file. FFmpeg reads the FLAC container, identifies the flac audio stream inside it, and fully decodes the losslessly compressed audio back into raw PCM samples before passing them to the output encoder. |
-c:a pcm_s16be
|
Sets the audio codec for the output to pcm_s16be — signed 16-bit big-endian PCM. This is the standard uncompressed audio encoding used inside AIFF files, producing CD-quality (or equivalent bit depth) audio with absolutely no compression or quality reduction. If your source FLAC is 24-bit, consider replacing this with pcm_s24be to preserve the full bit depth. |
output.aiff
|
Defines the output filename and format. The .aiff extension tells FFmpeg to use the AIFF container format, which wraps the pcm_s16be audio stream in Apple's Audio Interchange File Format structure, making it natively compatible with macOS applications like Logic Pro, GarageBand, and Final Cut Pro. |
Common Use Cases
- Importing audio into Apple Logic Pro or GarageBand, which natively prefer AIFF over FLAC for session tracks and audio clips
- Preparing lossless audio assets for Final Cut Pro timelines, where AIFF is the most seamlessly supported uncompressed audio format
- Delivering audio files to mastering engineers or studios working on macOS-based Pro Tools setups that expect AIFF rather than FLAC
- Converting a FLAC music archive to AIFF for playback on older iPod classic models or Apple hardware that does not support FLAC natively
- Creating AIFF source files for use in iOS audio apps or hardware samplers (such as the Akai MPC series) that read AIFF but not FLAC
- Uncompressing a FLAC file to AIFF before importing into Xcode for use as a sound effect asset in a macOS or iOS application
Frequently Asked Questions
No — this is a fully lossless conversion. FLAC is losslessly compressed PCM audio, and AIFF stores uncompressed PCM audio. FFmpeg decodes the FLAC samples and writes them directly into the AIFF container as pcm_s16be without any lossy encoding step. The audio waveform in the output AIFF file is bit-for-bit identical to what was stored in the FLAC source, assuming the FLAC was originally 16-bit (which is the most common case for CD-quality audio).
FLAC uses lossless compression — similar in concept to a ZIP file — to reduce file size while preserving all audio data. AIFF stores raw, uncompressed PCM samples with no compression whatsoever, so every audio sample takes its full amount of storage space. A typical 16-bit stereo CD-quality AIFF file consumes about 10 MB per minute of audio, whereas FLAC at compression level 5 might store the same audio in 4–6 MB per minute. Expect AIFF output files to be 1.5× to 3× larger than your FLAC source.
AIFF supports a limited subset of metadata via ID3 tags embedded in the file, but FLAC uses Vorbis Comment tags, which are a different metadata system. FFmpeg will attempt to map common fields like title, artist, album, and track number across to AIFF's ID3 container, and most standard tags will carry over successfully. However, FLAC-specific metadata such as replay gain values and cue sheet data will not be preserved in the AIFF output, as AIFF has no equivalent structures for those features.
Not with the default command shown on this page. The command uses -c:a pcm_s16be, which encodes to 16-bit PCM regardless of the source bit depth. If your FLAC source is 24-bit (common for high-resolution audio from studios or HDtracks), you should change the codec to pcm_s24be to preserve the full 24-bit depth. The FFmpeg command would become: ffmpeg -i input.flac -c:a pcm_s24be output.aiff. Using pcm_s16be on a 24-bit source will dither down to 16 bits, which is still excellent quality but is technically a slight reduction in dynamic range headroom.
FFmpeg itself does not support batch processing with a single command, but you can script it easily. On macOS or Linux, open Terminal and run: for f in *.flac; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.flac}.aiff"; done — this loops over every FLAC file in the current directory and produces a matching AIFF file. On Windows Command Prompt, use: for %f in (*.flac) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This is especially useful if your collection exceeds the 1 GB browser limit for individual files.
Yes — AIFF is one of Logic Pro's preferred native audio formats, and files converted with this tool will import cleanly into any Logic Pro project as audio regions or sample assets. Logic Pro reads pcm_s16be AIFF files natively without transcoding. If your FLAC source was 24-bit and you converted to pcm_s24be, Logic Pro will also recognize that bit depth correctly, which is useful for high-resolution recording projects.
Technical Notes
FLAC's compression is purely lossless — its compression_level parameter (0 through 8, defaulting to 5) only affects encoding speed and file size, never audio quality. When FFmpeg decodes a FLAC file, the output PCM samples are identical regardless of what compression level was used to create the FLAC. The AIFF format uses big-endian byte ordering for its PCM data, which reflects its origins on Motorola 68k and early PowerPC Apple hardware — this is why the codec is called pcm_s16be (signed 16-bit big-endian) rather than pcm_s16le (little-endian, as used in WAV files). AIFF and WAV contain the same type of PCM audio data but use different byte ordering and container structure, which is why macOS tools have historically favored AIFF while Windows tools favor WAV. AIFF does not support lossless compression, subtitles, multiple audio tracks, or chapters — it is a single-track uncompressed audio container. The AIFF format caps out at 4 GB due to its 32-bit chunk size fields, so extremely long high-sample-rate recordings could theoretically hit this ceiling, though it is rarely a practical issue for music-length files.