How to Convert FLAC to AIFF

FLAC and AIFF are both lossless, so converting between them does not reduce audio quality when you configure it correctly. FLAC compresses the audio to save space. AIFF stores uncompressed PCM, which is what many Apple apps, DAWs, DJ software and hardware samplers expect. This guide covers a free command-line method that works on Windows, macOS and Linux. It also explains the settings that most often cause problems: bit depth, metadata and album art.

Before you start

• Keep your original FLAC files until you have checked the converted copies. AIFF can always be converted back to FLAC losslessly, but a bad conversion cannot be undone if the source is gone.

• Check free disk space. AIFF files are uncompressed, so expect them to be noticeably larger than the FLAC files they came from.

• Find out the bit depth and sample rate of your source files, for example 16-bit/44.1 kHz or 24-bit/96 kHz. You need this to choose the right output setting. A media info tool or the ffprobe command shown later will tell you.

• Decide where the conversion should run. Desktop tools such as ffmpeg, Audacity or a dedicated audio converter process files locally on your computer. Web-based converters upload your audio to someone else's server. Before you use one, read its own terms about storage, retention and deletion, and avoid it for unreleased or private recordings.

• For the method below, install ffmpeg from your operating system's package manager or an official build for your platform. Then confirm that the command ffmpeg -version runs in a terminal.

Step-by-step instructions

These steps use ffmpeg because the commands are the same on every desktop platform and every setting is explicit. Open a terminal (Terminal on macOS or Linux, PowerShell or Command Prompt on Windows) in the folder that contains your FLAC files.

• Step 1: Convert a single file and keep 24-bit audio. Run: ffmpeg -i input.flac -c:a pcm_s24be -write_id3v2 1 output.aiff. The -c:a pcm_s24be option writes 24-bit big-endian PCM, which is the standard sample format for AIFF. The -write_id3v2 1 option asks ffmpeg to copy tags such as artist, album and title into the AIFF file.

• Step 2: Use 16-bit only when the source is 16-bit. If your FLAC is 16-bit, which is typical for CD rips, use -c:a pcm_s16be instead. Pick the codec on purpose: if you leave it out, ffmpeg's AIFF output defaults to 16-bit, which silently reduces a 24-bit source.

• Step 3: Leave the sample rate alone. ffmpeg keeps the original sample rate unless you add a resampling option such as -ar. Only resample if a specific device or project requires it, and remember that this changes the audio data.

• Step 4: Batch-convert a folder on macOS or Linux. Run: for f in *.flac; do ffmpeg -i "$f" -c:a pcm_s24be -write_id3v2 1 "${f%.flac}.aiff"; done. Each AIFF file is created next to its FLAC original with the same base name.

• Step 5: Batch-convert a folder in Windows PowerShell. Run: Get-ChildItem *.flac | ForEach-Object { ffmpeg -i $.FullName -c:a pcm_s24be -write_id3v2 1 ($.BaseName + '.aiff') }. The output files are written to the current folder.

• Step 6: If you prefer a graphical tool, open the FLAC file in an audio editor or converter, choose AIFF as the export format, and set the encoding to match the source bit depth. Menu names and default settings differ between applications and versions, so check the export dialog rather than assuming it preserves the bit depth.

Check the output

• Inspect the file's technical details. Run ffprobe -hide_banner output.aiff and confirm three things: the codec is pcm_s24be or pcm_s16be as intended, the sample rate matches the source, and the duration matches the original FLAC.

• Compare the same details against the source. Run ffprobe -hide_banner input.flac and check that the channel count and sample rate agree.

• Open the AIFF in the app or device you converted it for. Confirm that it plays from start to finish and that titles and artist names appear. If the target app shows no tags, see the next section.

• Treat the extensions .aif and .aiff as the same format. Some programs write the AIFF-C variant. Most modern software reads both, but older hardware may accept only plain AIFF, so check your device's documentation if a file is rejected.

• Delete the FLAC originals only after a sample of converted files plays correctly in your target app. Many people keep the FLAC versions as a compact archive instead.

Common problems and next steps

• Tags are missing in the AIFF. Make sure the command includes -write_id3v2 1. Some applications read AIFF metadata inconsistently, so if tags still do not appear, add them with a tag editor that supports ID3 in AIFF and recheck in the target app.

• Album art did not carry over. Embedded cover images are handled differently from audio streams, and results depend on the ffmpeg version and the source file. If artwork matters, add it afterwards with a tag editor rather than relying on the conversion command.

• The AIFF is 16-bit but the source was 24-bit. The output codec was not specified or was set to pcm_s16be. Convert again from the original FLAC with -c:a pcm_s24be.

• A sampler or older device will not load the file. The device may support only certain sample rates or bit depths. Check its specifications, and only then convert a copy with the matching -ar value and PCM codec.

• The command is not recognized. ffmpeg is either not installed or not on your system PATH. Reinstall it, or call it with its full path, then confirm that ffmpeg -version works.

• The converted files take up too much space. This is expected, because AIFF is uncompressed. If you need the space back, keep the FLAC files as your library and convert to AIFF only for the projects or devices that require it. You can also convert AIFF back to FLAC with ffmpeg -i input.aiff output.flac without losing audio quality.