MyMediaTools

Convert M4B to AIFF — Free Online Tool

Convert M4B audiobooks to AIFF, decoding AAC-compressed audio into uncompressed PCM (pcm_s16be) at full 16-bit depth. This is ideal when you need a lossless, editor-friendly master from an audiobook or podcast file without any further generational quality loss.

FFmpeg Command

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Estimated output:

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How It Works

M4B files store audio as AAC (Advanced Audio Coding), a lossy compressed format, wrapped in an MPEG-4 container that also carries chapter markers, bookmarks, and metadata. During this conversion, FFmpeg decodes the AAC audio stream entirely — decompressing it from its lossy encoded state back into raw PCM samples — and then re-encodes those samples into the pcm_s16be codec (big-endian signed 16-bit PCM), which is the native audio encoding used inside AIFF containers. Because AAC is lossy, the original compression artifacts cannot be reversed; the AIFF output will be a faithful, uncompressed representation of what the AAC decoder produces, but it will not recover detail discarded when the M4B was originally encoded. The AIFF container itself is lossless and uncompressed, so no additional quality is lost in the output stage. Chapter markers, bookmarks, and other M4B-specific metadata are not carried over, as the AIFF format does not support those structures.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg executable — the open-source multimedia processing engine that handles all decoding, encoding, and container remuxing. In the browser-based tool, this runs via FFmpeg.wasm compiled to WebAssembly; on the desktop, it runs as a native binary.
-i input.m4b Specifies the input file, which is an M4B audiobook container holding an AAC audio stream along with chapter markers and bookmarking metadata. FFmpeg reads and demuxes this MPEG-4 container to extract the compressed AAC audio for decoding.
-c:a pcm_s16be Sets the audio codec for the output to pcm_s16be — signed 16-bit big-endian PCM — which is the standard uncompressed audio encoding used inside AIFF files. This instructs FFmpeg to fully decode the AAC audio and re-encode it as raw 16-bit samples in big-endian byte order, the byte order required by the AIFF specification.
output.aiff Defines the output filename and container format. The .aiff extension tells FFmpeg to wrap the pcm_s16be audio stream in an Audio Interchange File Format container, Apple's lossless uncompressed audio format natively supported across all macOS and iOS audio software.

Common Use Cases

  • Importing an audiobook chapter into a professional audio workstation (DAW) like Logic Pro or Pro Tools for noise reduction, editing, or mastering, where AIFF is the preferred native format on macOS.
  • Archiving a purchased M4B audiobook as uncompressed AIFF so future re-encoding to any target format starts from the highest-fidelity decoded source rather than re-compressing already-compressed AAC.
  • Preparing podcast episode audio originally exported as M4B for submission to a radio broadcast or podcast network that requires uncompressed WAV or AIFF deliverables.
  • Splitting an M4B file into individual chapter AIFF files for use in a custom audiobook player or interactive audio application that cannot read MPEG-4 containers.
  • Analyzing the audio waveform of an audiobook file in a tool that only accepts uncompressed formats, such as for QA loudness checking or forensic audio review.
  • Feeding M4B narration audio into machine learning pipelines or speech recognition systems that require uncompressed PCM audio rather than AAC-encoded input.

Frequently Asked Questions

No — converting from M4B to AIFF does not recover or improve quality. The M4B file's audio is encoded in AAC, which is a lossy format that permanently discards some audio information during the original encoding. When FFmpeg decodes the AAC stream, it produces the best audio the decoder can reconstruct, but any detail lost during the original AAC compression is gone. The resulting AIFF is an uncompressed, lossless capture of that decoded output, meaning no further quality is lost going forward, but you are not gaining back anything that AAC removed.
M4B files use AAC compression, which achieves very small file sizes by discarding audio data that is perceptually less important. AIFF stores raw, uncompressed PCM audio — every sample is written in full with no compression applied. A typical audiobook M4B encoded at 128 kbps might expand to 10–15 times its original size as an AIFF file, since 16-bit stereo PCM at 44.1 kHz requires approximately 10 MB per minute of audio. This is expected and normal for any lossy-to-uncompressed conversion.
No. Chapter markers and bookmarks are features specific to the MPEG-4 container format used by M4B, and the AIFF format has no equivalent structure to store them. When FFmpeg writes the AIFF output, those structural metadata elements are discarded. If you need to preserve chapters, you would need to split the M4B into separate files per chapter before or during conversion, using FFmpeg's chapter-aware segment output options.
pcm_s16be stands for PCM signed 16-bit big-endian — it is uncompressed audio where each sample is stored as a 16-bit integer in big-endian byte order, which is the standard byte order for AIFF. 16-bit depth is standard CD quality and is appropriate for most audiobook and podcast source material, which is rarely mastered above 16-bit anyway. If your M4B was sourced from a 24-bit master and you need to preserve that depth, you can change the codec in the command to pcm_s24be for 24-bit AIFF output.
Replace pcm_s16be in the command with another supported AIFF codec. For 24-bit output, use -c:a pcm_s24be; for 32-bit integer, use pcm_s32be; or for 32-bit floating point, use pcm_f32be. The full command would look like: ffmpeg -i input.m4b -c:a pcm_s24be output.aiff. Keep in mind that since the M4B source is lossy AAC, increasing bit depth beyond 16-bit will increase file size without recovering any audio quality that AAC already discarded.
The command shown converts a single file, but you can adapt it for batch processing in a shell script. On macOS or Linux, use: for f in *.m4b; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4b}.aiff"; done. On Windows Command Prompt, use: for %f in (*.m4b) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This processes every M4B in the current directory and outputs a matching AIFF file. Batch processing in the browser tool requires converting files one at a time.

Technical Notes

The M4B format is a variant of M4A (and by extension MP4), distinguished primarily by its container-level support for chapter atoms and Apple's bookmarking system — both of which are silently dropped during AIFF output since AIFF has no equivalent metadata containers for these structures. The audio codec in virtually all M4B files is AAC, typically encoded at 64–128 kbps for audiobooks, which is well below the information density of uncompressed PCM. The FFmpeg conversion pipeline decodes the AAC bitstream using its native AAC decoder and writes big-endian signed 16-bit PCM samples into an AIFF container, which is compliant with Apple's original AIFF specification from 1989. Standard ID3-style tags such as title, author, and album may or may not survive the remux depending on FFmpeg version and the specific tags present; if metadata preservation is critical, pass -map_metadata 0 explicitly in the command. The output sample rate will match whatever the AAC stream used (typically 44100 Hz for audiobooks), as FFmpeg does not resample by default. AIFF files are natively readable on macOS by virtually all Apple software including iTunes/Music, GarageBand, Logic Pro, and QuickTime, making this format a reliable interchange target in Apple-centric audio workflows.

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