Convert ALAC to AIF — Free Online Tool
Convert ALAC audio files (.m4a) to AIF format by decoding Apple's lossless compressed codec into uncompressed PCM audio stored in Apple's AIFF container. This is a fully lossless conversion — every audio sample is preserved exactly, just decompressed from ALAC's compressed MPEG-4 wrapper into raw 16-bit big-endian PCM.
to
FFmpeg Command
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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
ALAC (Apple Lossless Audio Codec) stores audio as losslessly compressed PCM data inside an MPEG-4 container. During this conversion, FFmpeg decodes the ALAC bitstream back to its original raw PCM samples and repackages them as uncompressed 16-bit big-endian PCM (pcm_s16be) inside an AIFF container. Because ALAC is mathematically lossless, the decoded PCM samples are bit-for-bit identical to the original pre-compression audio — no quality is lost. The resulting AIF file will be significantly larger than the source .m4a because ALAC's compression (typically 40–60% size reduction over raw PCM) is removed. Note that AIFF does not support chapters, so any chapter markers embedded in the source ALAC file will be dropped during conversion.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing tool. In the browser-based version of this tool, FFmpeg runs as a WebAssembly binary (FFmpeg.wasm) entirely within your browser — no audio data is sent to any server. |
-i input.m4a
|
Specifies the input file — an MPEG-4 container (.m4a) holding an ALAC-encoded lossless audio stream. FFmpeg reads the ALAC codec box from the MPEG-4 file and prepares to decode it to raw PCM. |
-c:a pcm_s16be
|
Sets the output audio codec to signed 16-bit big-endian PCM, which is the standard uncompressed audio format used in AIFF files. This decodes the ALAC compression entirely, producing raw audio samples in the byte order required by the AIFF specification. |
output.aif
|
Specifies the output filename with the .aif extension, which signals FFmpeg to wrap the decoded pcm_s16be audio stream in an AIFF container. The .aif and .aiff extensions are interchangeable; both denote the Audio Interchange File Format. |
Common Use Cases
- Importing Apple Music lossless purchases into professional digital audio workstations like Pro Tools or Logic Pro that require uncompressed AIFF rather than compressed ALAC
- Preparing audio masters for CD duplication workflows where AIFF is the standard delivery format accepted by disc replication houses
- Working with older Mac audio software or hardware samplers (such as Akai or Roland) that support AIFF natively but cannot decode ALAC compression
- Archiving ALAC albums to AIFF for long-term preservation in environments where decompression software availability is uncertain — AIFF's uncompressed PCM requires no proprietary codec
- Delivering stems or sound design assets to collaborators or clients who work on non-Apple platforms where ALAC support is inconsistent but AIFF playback is universal in professional audio tools
- Preparing audio for vinyl mastering or broadcast delivery pipelines that mandate uncompressed AIFF as a technical specification
Frequently Asked Questions
No — this is a fully lossless conversion with zero quality loss. ALAC is a lossless compression format, meaning its compressed data decodes to the exact same PCM samples that were originally encoded. When FFmpeg decodes ALAC to pcm_s16be for AIFF, the resulting audio is mathematically identical to the source. The only caveat is bit depth: the default command encodes to 16-bit PCM, so if your ALAC source was recorded at 24-bit, you'll want to adjust the output codec to preserve that depth (see the FAQ about bit depth below).
The default FFmpeg command uses pcm_s16be, which is 16-bit PCM. If your ALAC source is 24-bit, you should change the audio codec flag to -c:a pcm_s24be to preserve full 24-bit depth in the output AIFF file. Running 'ffprobe input.m4a' beforehand will confirm your source bit depth. Downconverting a 24-bit ALAC to 16-bit AIF is still lossless in the sense that no lossy compression is applied, but you will lose the extra dynamic range headroom of the higher bit depth.
ALAC applies lossless compression to audio data, typically reducing file size by 40–60% compared to uncompressed PCM. When you convert to AIFF, you're removing that compression entirely — the output file contains raw, uncompressed PCM samples. A 50MB ALAC file might expand to 80–100MB as an uncompressed AIFF. This is expected and correct behavior; the audio content is identical, just stored without compression.
Basic ID3-style metadata tags (title, artist, album, track number) are generally preserved during the conversion because both MPEG-4 and AIFF support metadata chunks. However, embedded album artwork may not transfer reliably in all cases, and chapter markers embedded in the ALAC file will be dropped entirely since the AIFF format does not support chapters. If metadata preservation is critical, verify the output file's tags after conversion using a tool like MediaInfo or kid3.
On macOS or Linux, you can batch process an entire directory with a shell loop: 'for f in *.m4a; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4a}.aif"; done'. On Windows Command Prompt, use: 'for %f in (*.m4a) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif"'. This applies the same pcm_s16be conversion to every .m4a file in the current directory, outputting a matching .aif file for each. The browser-based tool processes files individually, so the FFmpeg command is especially useful for large batch jobs.
Despite being developed by Apple, AIFF is a widely supported format across all major platforms. VLC, Audacity, Adobe Audition, Ableton Live, and most professional DAWs on Windows and Linux read AIFF natively. The pcm_s16be codec used in this conversion is standard uncompressed PCM — the 'be' (big-endian) byte order is the AIFF standard and is handled transparently by all compliant audio software. You should encounter no compatibility issues on modern Windows or Linux audio applications.
Technical Notes
This conversion decodes ALAC's lossless compression algorithm (which uses rice coding and is stored in an 'alac' codec box inside an MPEG-4 container) and writes the resulting PCM data as pcm_s16be — signed 16-bit big-endian PCM — into an AIFF container. Big-endian byte order is a defining characteristic of AIFF (compared to WAV's little-endian PCM) and originates from the Motorola 68000 architecture of early Macintosh systems. The sample rate of the source file is preserved automatically by FFmpeg without any resampling. If your source ALAC was encoded at a high sample rate (88.2 kHz, 96 kHz, or 192 kHz), the output AIFF will maintain that rate. The primary technical consideration for this conversion is bit depth: ALAC supports 16, 20, 24, and 32-bit audio, but the default command outputs 16-bit. Users with high-resolution ALAC sources should explicitly specify -c:a pcm_s24be or -c:a pcm_s32be to avoid truncating bit depth. AIFF also has a theoretical maximum file size limit of 4GB (it uses a 32-bit chunk size field in its FORM/AIFF header), which at 24-bit/96kHz stereo corresponds to roughly 4 hours of audio — unlikely to be a practical concern for most use cases.