Extract Audio from M4V to AIF — Free Online Tool
Extract the audio track from an M4V video file and save it as a full-quality AIF file — uncompressed PCM audio stored in Apple's native lossless container. Because M4V typically carries AAC audio and AIF uses uncompressed PCM, the audio is decoded from lossy AAC and re-encoded to lossless PCM, giving you the highest-fidelity representation of the source audio possible.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
M4V files store audio as AAC (Advanced Audio Coding), a lossy compressed format. During this conversion, FFmpeg discards the video stream entirely using the -vn flag, then decodes the AAC audio from the M4V container and re-encodes it as 16-bit big-endian PCM (pcm_s16be) — the standard uncompressed audio format inside an AIF file. This is a lossy-to-lossless conversion in the sense that the output is technically uncompressed, but the quality ceiling is bounded by the original AAC encoding. No additional quality is lost during the conversion itself; you simply get the best possible reconstruction of the audio the AAC codec originally captured, stored without further compression in the AIF container.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. This is the same engine running inside the browser via WebAssembly — copying this command to your terminal will produce an identical AIF file from the same M4V input on your local machine. |
-i input.m4v
|
Specifies the input file — an M4V container, typically holding an H.264 or H.265 video stream and an AAC audio track. FFmpeg reads all streams from this file before applying any filtering or codec instructions. |
-vn
|
Disables video output entirely. Since AIF is a pure audio format, this flag tells FFmpeg to skip the video stream from the M4V rather than attempt to encode or include it, keeping the output file audio-only. |
-c:a pcm_s16be
|
Sets the audio codec to signed 16-bit big-endian PCM — the standard uncompressed audio encoding for AIF files. This decodes the lossy AAC audio from the M4V and stores it as raw, uncompressed samples in the native byte order expected by the AIF container. |
output.aif
|
Specifies the output filename with the .aif extension. FFmpeg uses this extension to automatically select the AIFF muxer, which wraps the pcm_s16be audio stream in the Audio Interchange File Format container structure that Apple applications expect. |
Common Use Cases
- Extracting the audio from an iTunes video download to import into a Mac-based DAW like Logic Pro or GarageBand, which work natively with AIF files
- Pulling a musical performance or concert recording from an M4V file to preserve it as an uncompressed archival audio master on a Mac system
- Extracting narration or voiceover audio from an iOS-produced M4V screencast to use as stems in an audio editing workflow
- Converting the audio from an Apple TV or iTunes M4V episode for use in a video essay or podcast, where PCM editing in AIF is preferred over compressed AAC
- Producing AIF source files from M4V content for mastering or audio restoration work, where compressed formats would degrade further with each processing pass
- Extracting soundtrack audio from M4V video files to load into Ableton Live or Pro Tools on Mac, where AIF is a preferred interchange format
Frequently Asked Questions
No. The M4V file's audio is encoded as AAC, which is a lossy format — some audio information was permanently discarded when the M4V was originally created. Converting to AIF with PCM audio faithfully decodes and stores what the AAC codec preserved, but it cannot recover data that was lost during the original AAC encoding. The result is an uncompressed, high-fidelity copy of the source audio with no additional quality loss introduced by this conversion.
AIF stores audio as raw, uncompressed PCM samples — 16-bit stereo audio at 44.1 kHz produces roughly 10 MB per minute, with no compression applied. M4V files store their audio as AAC, which typically achieves compression ratios of 10:1 or higher. Additionally, the M4V's video stream is removed, but the remaining uncompressed audio still far exceeds the size of the AAC-compressed audio track alone. This size increase is expected and is a direct consequence of moving from compressed to uncompressed audio storage.
Metadata preservation from M4V to AIF is limited. M4V files commonly store iTunes-style metadata (title, artist, album, artwork) using MP4 atoms, while AIF supports a MARK/NAME/ANNO chunk structure that is not fully compatible. FFmpeg does not reliably map M4V metadata to AIF format during this conversion, so you should expect most or all metadata to be absent in the output file. You can re-add metadata afterward using a tool like mp3tag or the Finder's Get Info panel on Mac.
Yes. M4V files can contain multiple audio tracks — for example, a primary mix and a secondary language track. By default, FFmpeg selects the first audio stream. To target a specific track, add -map 0:a:1 to the command to select the second audio stream (index 1), or -map 0:a:2 for the third, and so on. The full command would become: ffmpeg -i input.m4v -vn -map 0:a:1 -c:a pcm_s16be output.aif.
The default command uses pcm_s16be, which produces 16-bit AIF audio — standard CD quality. If you want higher bit depth, replace pcm_s16be with pcm_s24be for 24-bit audio or pcm_s32be for 32-bit audio. For example: ffmpeg -i input.m4v -vn -c:a pcm_s24be output.aif. Note that since the source AAC audio is already lossy, increasing bit depth will not recover lost information — it may, however, be appropriate for downstream mixing workflows that benefit from headroom in a 24-bit or 32-bit processing chain.
FFmpeg itself processes one file per command, but you can batch convert using a shell loop. On macOS or Linux, run: for f in *.m4v; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.m4v}.aif"; done. On Windows Command Prompt, use: for %f in (*.m4v) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aif". This will process every M4V in the current folder and produce a matching AIF file for each.
Technical Notes
The AIF format uses big-endian byte ordering, which is why the codec pcm_s16be (PCM signed 16-bit big-endian) is used rather than the little-endian pcm_s16le found in WAV files — this is a fundamental characteristic of AIF as an Apple format historically rooted in Motorola 68k architecture. The sample rate and channel layout of the output AIF will match whatever the source AAC track in the M4V contains — typically 44.1 kHz or 48 kHz stereo. FFmpeg does not resample or downmix by default. One important limitation: if the M4V file has DRM protection applied (as is common with purchased iTunes content), FFmpeg will be unable to read the audio stream, and the conversion will fail. Only DRM-free M4V files can be processed. Chapters and subtitle tracks present in the M4V are dropped automatically since AIF supports neither. If the source M4V uses libmp3lame as its audio codec rather than AAC (an atypical configuration), the same command applies — FFmpeg will decode MP3 audio and encode it to PCM just the same.