MyMediaTools

Convert AIFF to ALAC — Free Online Tool

Convert AIFF audio files to ALAC (Apple Lossless Audio Codec) stored in an M4A container — preserving every bit of audio data while applying lossless compression that Apple devices and iTunes natively support. AIFF's uncompressed PCM audio is re-encoded using the ALAC codec, typically reducing file size by 40–60% with zero quality loss.

FFmpeg Command

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How It Works

AIFF stores raw, uncompressed PCM audio (commonly 16-bit or 24-bit big-endian samples) with no compression whatsoever. During this conversion, FFmpeg decodes the PCM stream from the AIFF container and re-encodes it using the ALAC codec, which applies lossless compression — mathematically identical to the original but stored more efficiently. The output is wrapped in an MPEG-4 (.m4a) container rather than Apple's legacy AIFF container. Because both formats are lossless, the decoded audio at playback is bit-for-bit identical to the source — no frequency content, dynamic range, or sample precision is lost. The M4A container also supports chapter markers and richer metadata tagging than AIFF allows.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg application — the open-source multimedia processing engine that handles decoding the AIFF input and re-encoding it as ALAC. This tool runs FFmpeg compiled to WebAssembly (FFmpeg.wasm) directly in your browser.
-i input.aiff Specifies the input file — an AIFF file containing uncompressed PCM audio (typically pcm_s16be or pcm_s24be big-endian samples). FFmpeg reads the AIFF container and demuxes the raw audio stream for re-encoding.
-c:a alac Sets the audio codec for the output to ALAC (Apple Lossless Audio Codec). FFmpeg re-encodes the uncompressed PCM audio from the AIFF source into lossless ALAC compression, reducing file size while preserving audio data exactly.
-c:a alac A second instance of the ALAC codec flag as resolved by this tool's command builder — FFmpeg uses the final codec specification when duplicates are present, so the output is correctly encoded as ALAC. The behavior is identical to specifying it once.
output.m4a Defines the output filename with the .m4a extension, which signals FFmpeg to wrap the ALAC-encoded audio in an MPEG-4 container. The M4A container is the standard packaging for ALAC audio and is natively recognized by iTunes, Apple Music, iPhone, and iPad.

Common Use Cases

  • Shrinking a large AIFF recording session archive to ALAC to save disk space without any quality penalty before storing on an iPhone or iPad
  • Preparing high-resolution 24-bit studio recordings from a macOS DAW for import into iTunes or Apple Music library, which natively handles ALAC but treats AIFF as a bulkier alternative
  • Converting AIFF masters received from a recording studio into ALAC for distribution to audiophiles who want lossless audio compatible with Apple devices and AirPlay speakers
  • Reducing the size of AIFF sound design assets or sample libraries so they fit within tighter storage budgets on MacBook SSDs without re-recording or lossy compression
  • Migrating a legacy collection of AIFF files ripped from CDs on older Macs into a modern ALAC library that works seamlessly with Apple Music, HomePod, and Apple TV
  • Archiving broadcast-quality AIFF audio from video post-production into a smaller lossless format before long-term cold storage

Frequently Asked Questions

No — both AIFF and ALAC are lossless formats. AIFF stores raw PCM samples with no compression, while ALAC uses lossless compression to reduce file size. When FFmpeg decodes the AIFF and re-encodes to ALAC, the decompressed audio at playback is mathematically bit-for-bit identical to the original PCM data. No frequencies are removed, no dynamic range is altered, and no artifacts are introduced.
ALAC typically achieves a compression ratio of roughly 40–60% compared to uncompressed AIFF, depending on the audio content. Highly dynamic or complex material (like orchestral recordings) compresses less efficiently than sparse content (like spoken word or simple instrumentals). A 100 MB AIFF file will commonly become a 40–60 MB ALAC file. The exact reduction depends on bit depth, sample rate, and the nature of the audio signal.
FFmpeg will attempt to copy standard metadata tags from AIFF to the M4A container during conversion. However, AIFF uses ID3 or AIFF-specific chunk metadata, while M4A uses iTunes-style atom tags — so tag mapping is not always one-to-one. Common fields like title, artist, album, and track number generally carry over, but custom or non-standard tags may be dropped. It is worth verifying tags in iTunes or a tag editor like MusicBrainz Picard after conversion.
ALAC supports 16-bit, 20-bit, 24-bit, and 32-bit integer audio, so 16-bit and 24-bit AIFF files (the most common professional formats) convert cleanly with full bit depth preserved. AIFF files encoded with 32-bit or 64-bit floating-point PCM (pcm_f32be or pcm_f64be) require conversion to integer PCM before ALAC encoding, since ALAC does not support floating-point samples. FFmpeg handles this conversion automatically, but you should be aware that 64-bit float precision will be reduced to 32-bit integer depth.
Yes — ALAC in the M4A container is Apple's own lossless codec and is natively supported across the entire Apple ecosystem, including iPhone, iPad, Mac, Apple TV, HomePod, AirPlay 2, iTunes, and Apple Music. It is also supported in VLC, Windows Media Player (with appropriate codecs), and most modern Android music players. This makes ALAC a better long-term format choice than AIFF for Apple-centric workflows where broad device compatibility matters.
The single-file command shown here is 'ffmpeg -i input.aiff -c:a alac output.m4a'. To batch convert an entire folder on macOS or Linux, you can run: 'for f in *.aiff; do ffmpeg -i "$f" -c:a alac "${f%.aiff}.m4a"; done' in the terminal. On Windows Command Prompt, use: 'for %f in (*.aiff) do ffmpeg -i "%f" -c:a alac "%~nf.m4a"'. This processes each AIFF file in the current directory and outputs a matching ALAC M4A file with the same base filename.

Technical Notes

AIFF is a container format developed by Apple in 1988 that holds uncompressed PCM audio in big-endian byte order — the format stores every sample as-is with no compression, making files very large but requiring minimal CPU to decode. ALAC, introduced by Apple in 2004 and open-sourced in 2011, uses a predictive coding algorithm similar in concept to FLAC but tuned for Apple's ecosystem and stored in the MPEG-4 container. One technical consideration is that the AIFF format supports floating-point PCM (pcm_f32be and pcm_f64be), while ALAC only supports integer PCM up to 32 bits. FFmpeg will automatically convert floating-point samples to integer before ALAC encoding, which may introduce negligible rounding differences at the extreme low end of the dynamic range — well below audible thresholds. The M4A container supports chapter markers (which AIFF does not), so if you add chapters after conversion, ALAC M4A can accommodate them. Both formats are natively recognized by macOS without third-party software. Note that the FFmpeg command includes '-c:a alac' twice due to how this tool resolves codec flags — FFmpeg processes only the final codec specification, so the output is correctly encoded as ALAC regardless.

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