Convert M4B to AIF — Free Online Tool
Convert M4B audiobook files to AIF format, decoding the AAC-encoded audio stream into uncompressed PCM data for use in professional audio applications on Mac. This is ideal when you need a lossless-quality master of an audiobook chapter for editing in Logic Pro or other DAWs that work natively with Apple's AIF format.
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FFmpeg Command
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Estimated output:
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DownloadHow 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 podcast metadata. During conversion, FFmpeg decodes the AAC audio stream back into raw PCM samples and then re-encodes them as big-endian 16-bit PCM (pcm_s16be), writing the result into an AIF container. Because AAC is lossy, the conversion is a lossy-to-lossless process: the AIF output is uncompressed and will never degrade further, but the audio fidelity is bounded by the quality of the original AAC encoding. Chapter markers, bookmarking metadata, and podcast tags carried in the M4B container are not preserved in AIF, as the format does not support those features.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, which handles the decoding of the AAC audio inside the M4B container and the encoding of the output AIF file. |
-i input.m4b
|
Specifies the input M4B audiobook file. FFmpeg reads the MPEG-4 container, identifies the AAC audio stream, and prepares it for decoding. |
-c:a pcm_s16be
|
Sets the audio codec for the output to 16-bit big-endian signed PCM, which is the standard uncompressed audio encoding used in Apple's AIF format. This decodes the lossy AAC audio from the M4B into raw, uncompressed samples. |
output.aif
|
Defines the output filename and tells FFmpeg to write an AIF container. The .aif extension causes FFmpeg to use the AIF muxer, which packages the pcm_s16be audio stream in the big-endian AIFF structure expected by macOS audio applications. |
Common Use Cases
- Loading an audiobook chapter into Logic Pro or GarageBand for audio restoration, noise reduction, or EQ processing before archiving
- Importing an M4B podcast episode into a professional DAW to extract and re-edit specific interview segments as uncompressed audio
- Creating an uncompressed archival copy of a purchased audiobook so it can be re-encoded later at any bitrate without generation loss
- Using an M4B narration file as a source asset in a video production timeline where the NLE requires uncompressed AIF audio
- Providing a lossless AIF file to an audio mastering engineer who needs a workable source even if the original recording was AAC-compressed
- Converting M4B spoken-word content into AIF for playback on legacy Mac hardware or software that does not support MPEG-4 audio containers
Frequently Asked Questions
No. The M4B stores audio as AAC, which is a lossy format — some audio information was discarded when the M4B was originally created. Converting to AIF decodes the AAC and stores the result as uncompressed PCM, which is technically lossless, but it cannot recover data that AAC already discarded. The AIF output is a perfect, non-degrading copy of whatever fidelity remains after the AAC encoding, not a higher-quality master.
AAC compression in M4B files typically reduces audio data to around 10–20% of its uncompressed size. AIF stores audio as raw uncompressed PCM samples with no compression at all — a 1-hour M4B at 128k AAC (roughly 57 MB) will expand to approximately 600 MB as a 16-bit stereo AIF at 44.1 kHz. This size increase is expected and reflects the difference between compressed and uncompressed audio storage.
No. The AIF format does not support chapter markers, bookmarks, or the podcast and audiobook metadata fields (such as author, narrator, or chapter titles) that M4B carries. All of that structural information is lost during conversion. If you need to retain chapter data, you should export individual chapters as separate M4B or AAC files before converting to AIF.
pcm_s16be stands for 16-bit signed PCM with big-endian byte ordering, which is the standard PCM format for AIF files on Apple systems. It provides a dynamic range of about 96 dB, which is more than sufficient for audiobook and spoken-word content. If you are working with high-resolution audio sources or need to match a 24-bit studio workflow, you could modify the FFmpeg command to use pcm_s24be instead, though the extra bit depth will not recover detail lost in the original AAC encoding.
On macOS or Linux you can use a shell loop: `for f in *.m4b; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.m4b}.aif"; done`. On Windows Command Prompt, use: `for %f in (*.m4b) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif"`. Each file is processed sequentially, decoded from AAC and written out as an uncompressed AIF.
Yes. Replace `pcm_s16be` in the command with `pcm_s24be` for 24-bit or `pcm_s32be` for 32-bit output: `ffmpeg -i input.m4b -c:a pcm_s24be output.aif`. Higher bit depths produce larger files and are more relevant when working with high-resolution source material; for typical audiobook M4B files encoded at 64–128k AAC, 16-bit is sufficient and keeps file sizes manageable.
Technical Notes
The core technical challenge of this conversion is that it crosses the lossy-to-lossless boundary: AAC in M4B is a perceptual codec that permanently removes audio data deemed inaudible, so the resulting AIF is lossless in storage but not in origin. The default output codec pcm_s16be is big-endian 16-bit PCM, the canonical encoding for Apple's AIF container and natively supported by every macOS audio application. M4B files may contain multiple audio tracks in rare cases, but this command processes only the default (first) audio stream; if your file has alternate language tracks you would need to specify `-map 0:a:1` explicitly. The M4B container's -movflags +faststart optimization, chapter atom (chpl), and iTunes metadata atoms (©nam, ©ART, etc.) are all silently discarded since AIF has no equivalent structures for them. Sample rate is passed through unchanged from the source — most M4B audiobooks are 44.1 kHz or 22.05 kHz — so no resampling occurs unless explicitly requested. File sizes will typically increase by a factor of 8–12x compared to the source M4B depending on the original AAC bitrate.