Convert M4B to AIFC — Free Online Tool
Convert M4B audiobooks to AIFC format, transcoding AAC-encoded audio to uncompressed PCM (pcm_s16be) for use in professional audio workflows and legacy Apple applications. This tool runs entirely in your browser — no uploads required — and displays the exact FFmpeg command for local processing.
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
M4B files store audio as AAC (Advanced Audio Coding), a lossy compressed format optimized for audiobook delivery with support for chapters and bookmarks. AIFC, the compressed/extended variant of Apple's AIFF format, can store both compressed and uncompressed audio; in this conversion, FFmpeg decodes the AAC stream from the M4B container and re-encodes it as 16-bit big-endian PCM (pcm_s16be) inside the AIFC container. This means the audio is fully decoded from its lossy AAC representation and written as raw pulse-code modulation data — a fundamentally different encoding pathway. M4B metadata like chapters, bookmarks, and audiobook tags are not preserved in AIFC, as the format does not support these features. The resulting file will be significantly larger than the original M4B because PCM audio is uncompressed.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing engine. In the browser-based tool, this runs via FFmpeg.wasm (WebAssembly) entirely client-side; when run locally, this calls your desktop FFmpeg installation. |
-i input.m4b
|
Specifies the input file — an M4B audiobook container holding an AAC-encoded audio stream along with chapter markers and bookmarking metadata. FFmpeg will demux the AAC audio from the MPEG-4 container for decoding. |
-c:a pcm_s16be
|
Decodes the source AAC audio and re-encodes it as signed 16-bit big-endian PCM, the native uncompressed audio format expected by AIFC containers. Big-endian byte order is required by the AIFC specification, distinguishing it from the little-endian PCM used in WAV files. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps, inherited from the tool's general quality parameter. This flag has no practical effect on uncompressed PCM codecs like pcm_s16be — PCM bitrate is determined entirely by sample rate and bit depth — so it is effectively a no-op in this command. |
output.aifc
|
Defines the output filename with the .aifc extension, which signals FFmpeg to write an Audio Interchange File Format Compressed container. The AIFC container wraps the decoded pcm_s16be audio stream along with any basic metadata fields the format supports. |
Common Use Cases
- Importing an audiobook's audio track into a professional DAW like Pro Tools or Logic Pro, which expects uncompressed AIFF/AIFC source files for editing or mastering
- Archiving a purchased M4B audiobook as uncompressed PCM audio to ensure long-term playback compatibility independent of AAC decoder support
- Preparing M4B-sourced narration audio for post-production processing — such as noise reduction or normalization — in tools that require uncompressed input
- Providing a chapter segment extracted from an M4B to a legacy Mac application or hardware device that reads AIFC but not AAC-in-MPEG4 containers
- Converting podcast episodes delivered in M4B format to uncompressed AIFC for broadcast playout systems that require PCM audio
Frequently Asked Questions
No — the audio quality is capped at whatever was captured in the original AAC encoding of the M4B. When FFmpeg decodes the AAC stream, any information discarded during the original lossy compression is already gone permanently. The resulting AIFC file stores those decoded samples in uncompressed PCM, which is a lossless representation of the decoded output, but it cannot recover detail lost during the original AAC encoding. Think of it as a perfect copy of an imperfect source.
AAC, used in M4B files, is a lossy compression codec that typically achieves 10:1 or greater compression compared to uncompressed audio. The AIFC output uses 16-bit big-endian PCM (pcm_s16be), which stores every audio sample as raw numerical data with no compression. A typical M4B audiobook chapter compressed at 128 kbps AAC might expand to roughly 1.4 MB per minute in AIFC at CD-quality sample rates, compared to under 1 MB per minute in the M4B. For full-length audiobooks, expect the AIFC file to be 8–12 times larger.
No. M4B's chapter markers, bookmarking positions, and audiobook metadata tags are embedded in the MPEG-4 container structure and have no equivalent in the AIFC format. When FFmpeg writes the AIFC output, it only carries over basic metadata that AIFC supports, such as title and artist fields. If you need to preserve chapter structure, you should split the M4B into per-chapter audio files before converting each segment to AIFC.
pcm_s16be stands for PCM (pulse-code modulation), signed 16-bit samples, big-endian byte order — the same bit depth and byte order used by standard Audio CDs and classic Mac audio. It is the default and most compatible PCM variant for AIFC. If your DAW or application requires higher bit depth, you could modify the FFmpeg command to use pcm_s24be (24-bit) or pcm_s32be (32-bit) instead of pcm_s16be, though the source AAC audio was almost certainly mastered at 16-bit or lower effective resolution anyway.
To change the PCM bit depth, replace pcm_s16be with another supported AIFC codec such as pcm_s24be or pcm_s32be in the -c:a flag. To resample the audio to a different sample rate, add the flag -ar followed by the target rate — for example, -ar 44100 for CD quality or -ar 48000 for broadcast standard — inserted before the output filename. A full command targeting 24-bit at 48 kHz would look like: ffmpeg -i input.m4b -c:a pcm_s24be -ar 48000 output.aifc. Note that the -b:a bitrate flag has no meaningful effect on uncompressed PCM codecs; file size is determined entirely by bit depth, sample rate, and duration.
Yes. On Linux or macOS, you can loop over all M4B files in a directory with a shell one-liner: for f in *.m4b; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4b}.aifc"; done. On Windows Command Prompt, use: for %f in (*.m4b) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc". This is especially useful when you have an audiobook split across multiple M4B files per chapter and want to convert each to a discrete uncompressed AIFC file for DAW import.
Technical Notes
M4B is an MPEG-4 container variant (essentially a renamed .m4a) distinguished by its support for chapter markers, Apple's bookmarking extension, and AAC audio encoding — features that make it purpose-built for audiobook distribution. AIFC (Audio Interchange File Format Compressed) is Apple's extension of AIFF that accommodates both uncompressed PCM and compressed audio codecs, though in practice the format is most commonly used with PCM variants in professional and archival contexts. The default codec in this conversion, pcm_s16be, uses big-endian byte ordering as AIFC requires, which differs from the little-endian PCM used in WAV files — the two formats are not byte-for-byte interchangeable even at the same bit depth and sample rate. Because the source AAC audio is lossy, the conversion introduces no additional generation loss beyond what already existed, but it also recovers none of it; the AIFC file is a faithful lossless container for the post-decode PCM samples. The -b:a 128k flag in the command is carried over from general tool scaffolding but has no effect on uncompressed PCM codecs — PCM bitrate is fixed by sample rate and bit depth, not a compression parameter. AIFC files produced by this conversion will be fully readable by macOS's Core Audio framework, Logic Pro, GarageBand, and any application using the standard AIFF/AIFC parser, but will not be recognized as audiobooks by any player since AIFC carries no audiobook metadata schema.