Extract Audio from MP4 to AIF — Free Online Tool
Extract audio from an MP4 video and save it as a lossless AIF file using PCM 16-bit big-endian encoding — ideal for Mac-based audio workflows that require uncompressed, studio-quality audio. Unlike lossy extraction to MP3 or AAC, this conversion preserves every bit of the original audio signal with no re-encoding artifacts.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your MP4 file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
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
During this conversion, FFmpeg discards the video stream entirely and extracts the audio track from the MP4 container. Because MP4 typically stores audio as compressed AAC (or sometimes MP3 or Opus), the audio must be fully decoded from its compressed format and then re-encoded as uncompressed PCM 16-bit big-endian (pcm_s16be) — the native codec for AIF files. This is a decode-then-encode operation, not a simple remux. The result is a lossless AIF file where audio samples are stored in raw uncompressed form, making it significantly larger than the original AAC audio but free of any lossy compression. The big-endian byte order used by pcm_s16be reflects AIF's Apple origins and is natively compatible with macOS audio tools and digital audio workstations.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program — the open-source multimedia processing engine running as WebAssembly in your browser for this tool, or available as a desktop command-line application for local use with files over 1GB. |
-i input.mp4
|
Specifies the input file — an MP4 container that typically holds a video stream (often H.264) and a compressed audio stream (usually AAC), both of which FFmpeg will read and process. |
-vn
|
Disables video output entirely, telling FFmpeg to discard the H.264 (or other) video stream from the MP4. Since AIF is a audio-only format, this flag ensures no video data is passed through to the output. |
-c:a pcm_s16be
|
Sets the audio codec to PCM 16-bit signed big-endian, which is the standard uncompressed audio encoding used in AIF files. This decodes the compressed AAC audio from the MP4 and re-encodes it as raw uncompressed samples in the byte order native to the AIF format. |
output.aif
|
Specifies the output filename with the .aif extension, which tells FFmpeg to write an Audio Interchange File Format container — a format developed by Apple for storing uncompressed PCM audio, natively compatible with macOS audio applications and DAWs. |
Common Use Cases
- Extracting a music performance or recorded audio from an MP4 video to import into Logic Pro or GarageBand as a lossless AIF session file
- Pulling clean, uncompressed dialogue audio from an MP4 interview recording for professional audio editing or podcast post-production on a Mac
- Converting MP4 video game captures or film clips into AIF for use as sound design source material in a DAW that prefers uncompressed audio
- Archiving the audio track of an important video in an uncompressed format to avoid any further generational quality loss in future re-encodings
- Extracting audio from an MP4 training video or lecture to create an AIF source file for transcription software that requires high-fidelity, uncompressed input
- Preparing audio from an MP4 music video for mastering by delivering an uncompressed AIF file to an audio engineer who works in the Apple ecosystem
Frequently Asked Questions
No — and this is an important nuance. The audio in most MP4 files is already stored as compressed AAC, which is a lossy format. When FFmpeg decodes that AAC audio and re-encodes it as uncompressed PCM in the AIF file, no lost information is recovered. The AIF file will be lossless and uncompressed going forward, but it cannot restore quality that was discarded when the AAC stream was originally encoded. What it does do is prevent any further quality degradation if you plan to re-encode or process the audio again in a DAW.
AIF stores audio as raw, uncompressed PCM samples — every audio sample is written directly to disk with no compression applied. AAC audio in an MP4 file, by contrast, achieves compression ratios of roughly 10:1 or higher. A 128 kbps AAC audio track extracted as 16-bit PCM AIF at 44.1 kHz stereo will require approximately 1.4 MB per minute per channel, totalling around 10 MB per minute for stereo — versus roughly 1 MB per minute for the AAC original. The video stream being removed does reduce the file size somewhat, but the uncompressed audio format more than compensates.
This tool uses pcm_s16be — 16-bit signed big-endian PCM — which is the standard and most compatible AIF codec. AIF also supports higher bit-depth codecs such as pcm_s24be (24-bit) and pcm_s32be (32-bit), which are preferred in professional audio production. However, since the source audio in the MP4 is almost certainly AAC at effectively 16-bit fidelity or less, using pcm_s16be is appropriate and avoids unnecessary file size inflation from a higher bit depth that would not contain additional real information.
Metadata compatibility between MP4 and AIF is limited. MP4 uses iTunes-style metadata tags (title, artist, album, artwork, etc.) stored in the moov atom, while AIF uses MARK, NAME, and ID3-style chunks for metadata. FFmpeg will attempt to map common tags such as title and artist, but embedded album art and some MP4-specific metadata fields are not carried over. If preserving rich metadata is important, you should verify the output AIF file's tags in a tool like iTunes or Kid3 after conversion.
Yes. To output 24-bit AIF instead of 16-bit, replace pcm_s16be with pcm_s24be in the command: ffmpeg -i input.mp4 -vn -c:a pcm_s24be output.aif. This is useful if your downstream workflow specifically requires 24-bit files, though keep in mind that 24-bit will not recover quality lost in the original AAC compression. For 32-bit float output, you can use pcm_f32be, which some DAWs use internally for processing.
Yes. On macOS or Linux, you can use a shell loop to process multiple files at once: for f in *.mp4; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mp4}.aif"; done. On Windows using PowerShell, you can achieve the same with: Get-ChildItem *.mp4 | ForEach-Object { ffmpeg -i $_.FullName -vn -c:a pcm_s16be ($_.BaseName + '.aif') }. Batch processing via the command line is especially useful for large files over 1GB, which exceed the browser tool's file size limit.
Technical Notes
AIF (Audio Interchange File Format) was developed by Apple in 1988 and uses big-endian byte ordering inherited from the Motorola 68k architecture of early Macintosh systems — which is why the codec is designated pcm_s16be (signed 16-bit big-endian) rather than the little-endian pcm_s16le used in WAV files. This byte order distinction matters when exchanging files between systems, though all major DAWs and audio tools on both Mac and Windows handle AIF correctly. One important limitation of AIF is that it does not support multiple audio tracks — only the first (or default) audio stream from the MP4 will be extracted. If your MP4 contains multiple audio tracks (e.g., multiple language tracks), you can select a specific one using the -map flag in the FFmpeg command, such as -map 0:a:1 to select the second audio track. AIF also does not support embedded subtitle tracks or chapter markers, so those will be silently dropped. The resulting AIF file will be compatible with all Apple applications including Logic Pro, GarageBand, Final Cut Pro, and iTunes, as well as cross-platform DAWs like Ableton Live, Pro Tools, and Reaper.