Convert ALAC to AIFF — Free Online Tool
Convert ALAC (Apple Lossless Audio Codec) files stored in M4A/MP4 containers to AIFF format by decoding the losslessly compressed audio into uncompressed PCM (16-bit big-endian) samples. Both formats are lossless and Apple-native, so this conversion preserves every detail of your original audio with zero quality loss.
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FFmpeg Command
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Estimated output:
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DownloadHow It Works
ALAC stores audio as losslessly compressed data inside an MPEG-4 container — the audio is mathematically identical to the original source, but takes up less disk space due to compression. AIFF, by contrast, stores audio as raw, uncompressed PCM samples (in this case, 16-bit big-endian, matching the bit depth typical of CD-quality audio). During conversion, FFmpeg fully decodes the ALAC stream into raw PCM audio frames and writes them into an AIFF container using the pcm_s16be codec. Because both formats are lossless, no audio information is discarded — the decoded waveform is identical to what ALAC would produce. The main practical effect is a significant increase in file size, since AIFF carries no compression at all, while ALAC typically achieves 40–60% compression over uncompressed equivalents.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, the open-source multimedia processing engine that handles decoding the ALAC stream from the M4A container and re-encoding it as uncompressed PCM inside an AIFF file. |
-i input.m4a
|
Specifies the input file — an M4A file containing an ALAC-encoded audio stream. FFmpeg reads the MPEG-4 container, locates the ALAC audio track, and begins decoding it into raw PCM frames. |
-c:a pcm_s16be
|
Sets the audio codec for the output to pcm_s16be — signed 16-bit PCM with big-endian byte ordering. This is the standard uncompressed audio encoding for AIFF files, equivalent to CD-quality audio, and is what AIFF players and DAWs expect when reading the format. |
output.aiff
|
Defines the output filename and, through its .aiff extension, instructs FFmpeg to wrap the uncompressed PCM audio in an Audio Interchange File Format (AIFF) container — the standard Apple uncompressed audio format compatible with macOS, Logic Pro, and professional audio software. |
Common Use Cases
- Bringing Apple Music library files (downloaded as ALAC/M4A) into professional digital audio workstations like Pro Tools or Logic Pro that work best with uncompressed AIFF session files.
- Preparing audio assets for mastering engineers or studios whose workflows require uncompressed AIFF files rather than compressed containers.
- Archiving lossless music on macOS systems in the traditional AIFF format favored by older Apple software and hardware samplers like the Akai MPC series.
- Converting ALAC podcast or voice recordings to AIFF for editing in audio software that does not natively decode ALAC within M4A containers.
- Importing music into DJ software such as Serato or Traktor that indexes AIFF files more reliably than ALAC-in-M4A for BPM analysis and waveform display.
- Exporting ALAC source files to AIFF to use as audio sources in video editing timelines where AIFF is the preferred uncompressed format for frame-accurate sync.
Frequently Asked Questions
No — this conversion is entirely lossless. ALAC is a lossless compression format, meaning the decompressed audio data is bit-for-bit identical to the original uncompressed source. AIFF stores that same audio as raw PCM samples without any further processing. The waveform you hear from the AIFF output is mathematically identical to what the ALAC file contains, just no longer compressed.
ALAC uses lossless compression to reduce file size — typically achieving 40–60% smaller files compared to uncompressed audio — while preserving every sample perfectly. AIFF stores audio as raw, uncompressed PCM data with no compression whatsoever, so it takes up the full uncompressed size. For example, a 30-minute stereo ALAC file at 44.1kHz/16-bit might be around 150MB, while the equivalent AIFF could be 300MB or more. The audio content is identical; only the storage method differs.
pcm_s16be stands for signed 16-bit PCM with big-endian byte order, which is the standard PCM encoding used in AIFF files and matches CD-quality audio (44.1kHz, 16-bit). If your source ALAC was recorded or encoded at a higher bit depth — such as 24-bit for studio masters — you can preserve that depth by changing the codec flag to '-c:a pcm_s24be' in the command. This avoids any quantization from 24 to 16 bits and keeps your high-resolution audio fully intact.
Partially. ALAC files in M4A containers can carry rich iTunes-style metadata tags (artist, album, artwork, etc.). AIFF supports a basic set of metadata via its MARK and NAME chunks, and FFmpeg will attempt to carry over common tags. However, AIFF's metadata support is more limited than M4A's, and some fields — such as embedded album artwork — may not transfer completely. If metadata preservation is critical, verify the output file in your audio player or tag editor after conversion.
Yes. The default command uses '-c:a pcm_s16be' for 16-bit AIFF output, but AIFF supports higher bit depths. To output 24-bit AIFF (ideal for studio-quality source files), replace the codec flag with '-c:a pcm_s24be'. For 32-bit float output, use '-c:a pcm_f32be'. The full command for 24-bit would be: ffmpeg -i input.m4a -c:a pcm_s24be output.aiff. Check what bit depth your original ALAC was encoded at and match it accordingly to avoid unnecessary downsampling.
Yes. On macOS or Linux, you can loop over all M4A files in a directory using a shell one-liner: for f in *.m4a; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4a}.aiff"; done. On Windows Command Prompt, use: for %f in (*.m4a) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This is especially useful for large collections or files over 1GB that exceed the browser tool's file size limit.
Technical Notes
ALAC and AIFF are both lossless, Apple-originated formats, but they differ fundamentally in storage approach: ALAC applies entropy coding (similar in principle to FLAC) to reduce file size without losing data, while AIFF holds raw integer or float PCM samples with no compression. The FFmpeg command decodes ALAC using the native alac decoder and re-encodes the output as pcm_s16be — signed 16-bit big-endian PCM — which is the canonical audio encoding for AIFF and matches the byte order expected by the format's Apple/Motorola heritage. One important consideration is bit depth: if your ALAC source was mastered at 24-bit (common for high-resolution audio purchases), the default pcm_s16be output will truncate to 16 bits, losing some dynamic range headroom. Use pcm_s24be in that case. Chapter markers stored in the ALAC/M4A container will not be carried over, as AIFF does not support chapter data. The AIFF format has no native support for multiple audio tracks, matching the single-track nature of ALAC-in-M4A. Sample rate is preserved exactly as-is; FFmpeg does not resample unless explicitly instructed, so a 48kHz ALAC source will produce a 48kHz AIFF output.