MyMediaTools

Extract Audio from MP4 to AIFF — Free Online Tool

Extract audio from an MP4 video and save it as a lossless AIFF file using PCM 16-bit big-endian encoding — the native uncompressed format developed by Apple. Unlike lossy extraction to MP3 or AAC, this conversion preserves every sample from the source audio stream at full fidelity, making it ideal for professional audio work on macOS.

FFmpeg Command

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Estimated output:

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How It Works

MP4 files most commonly store audio in AAC (Advanced Audio Coding) format, which is a lossy compressed codec. When converting to AIFF, the AAC audio stream is fully decoded from its compressed state back into raw PCM audio samples, then re-encoded as 16-bit big-endian PCM (pcm_s16be) — the standard AIFF codec. This is a decode-and-re-encode process, not a remux. Because AAC is lossy, the original compression artifacts are already baked into the audio; the AIFF output captures those samples exactly without introducing any further quality degradation. The video stream is entirely discarded using the -vn flag, so no video processing occurs. The result is an uncompressed AIFF file that is significantly larger than the source MP4 audio but fully compatible with macOS audio tools, Logic Pro, GarageBand, and professional DAWs.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg command-line tool. In the browser, this runs via FFmpeg.wasm compiled to WebAssembly, so the same command works identically in a terminal on macOS, Linux, or Windows.
-i input.mp4 Specifies the input file — an MP4 container that typically holds a video stream (often H.264) and an audio stream (most commonly AAC). FFmpeg reads both streams but will only process the audio in this command.
-vn Disables video output entirely, telling FFmpeg to ignore the video stream from the MP4. This is necessary because AIFF is a pure audio container and cannot hold video data — without this flag, FFmpeg would attempt to map the video stream and fail.
-c:a pcm_s16be Sets the audio codec to 16-bit signed big-endian PCM, which is the standard uncompressed audio encoding used in AIFF files. The AAC audio from the MP4 is fully decoded and written out as raw PCM samples in big-endian byte order, conforming to Apple's AIFF specification.
output.aiff Specifies the output filename with the .aiff extension, which tells FFmpeg to use the AIFF muxer to wrap the PCM audio data. The resulting file is a standard AIFF that can be opened natively by macOS Finder, Logic Pro, GarageBand, and any DAW with AIFF support.

Common Use Cases

  • Extracting a music track from an MP4 music video download to import into Logic Pro or Pro Tools as a lossless AIFF for mixing or mastering
  • Pulling a recorded voiceover or narration from a video file into AIFF format for use as a source asset in a macOS audio post-production workflow
  • Archiving the audio from a recorded live performance or concert video in an uncompressed format before the original MP4 is deleted or archived to cold storage
  • Preparing audio content recorded in an MP4 screen capture for further processing in GarageBand, where AIFF is a natively preferred import format
  • Extracting dialogue or sound effects from an MP4 video clip for use as uncompressed source samples in a sound design library on macOS
  • Converting an MP4 audiobook or lecture recording to AIFF so it can be losslessly edited, trimmed, and re-exported without any generation loss in an Apple-native workflow

Frequently Asked Questions

No — the AIFF output will not sound better than the audio in the source MP4. Most MP4 files store audio as AAC, which is a lossy format, meaning some audio data was discarded when the MP4 was originally encoded. Converting to AIFF decodes those AAC samples into uncompressed PCM, but it cannot recover the detail that was lost during the original AAC encoding. What you gain is a lossless container for those existing samples, with no further quality degradation from this point forward in your workflow.
AIFF stores audio as raw, uncompressed PCM data with no compression applied. A standard stereo AIFF at 44.1 kHz with 16-bit depth uses roughly 10 MB per minute of audio. MP4 files typically store audio as AAC, which achieves compression ratios of 10:1 or more. So an MP4 with a few minutes of audio might have a tiny audio track, while the equivalent AIFF will be dramatically larger. The video stream is also stripped out entirely, so the AIFF contains only audio data.
AIFF supports a limited subset of metadata through its ID3 and MARK/NAME chunk mechanisms, but metadata from MP4 containers (stored as iTunes-style atoms) is not automatically mapped or preserved during this FFmpeg conversion. Fields like title, artist, album, and cover art will likely be lost in the output AIFF. If metadata preservation is important, you should use a dedicated tagging tool such as Kid3 or mp3tag after the conversion to manually re-apply the relevant tags to the AIFF file.
The FFmpeg command uses pcm_s16be, which produces 16-bit signed big-endian PCM — the default AIFF codec. To get 24-bit output, replace -c:a pcm_s16be with -c:a pcm_s24be in the command. For 32-bit integer PCM use pcm_s32be, and for 32-bit floating point use pcm_f32be. Higher bit depths produce larger files but can preserve more headroom if your source audio was recorded at 24-bit or higher. If your source MP4 contains AAC audio, which is inherently 16-bit after decoding, outputting at 24-bit will not add real audio information but may be required by certain DAW import workflows.
The single-file command shown here processes one file at a time, but you can adapt it for batch processing in a shell script. On macOS or Linux, you can run: for f in *.mp4; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mp4}.aiff"; done — this loops over every MP4 in the current directory and produces a corresponding AIFF file. On Windows using Command Prompt, a similar for loop can be constructed with a .bat script. This is especially useful for files over 1GB, which exceed the browser tool's limit but run efficiently on the desktop.
AIFF was developed by Apple and is most natively supported on macOS, but it is not exclusively a macOS format. Windows DAWs like Reaper, Adobe Audition, and Steinberg Cubase all support AIFF import and export. However, Windows Media Player and many common Windows media players do not support AIFF natively. If you need broad Windows compatibility for uncompressed audio, WAV (using pcm_s16le with little-endian byte order) is typically a better choice, as it is the Windows-native equivalent of AIFF.

Technical Notes

The default codec selected for this conversion is pcm_s16be — signed 16-bit PCM with big-endian byte ordering, which is the canonical AIFF audio format as defined by Apple's original specification. Big-endian byte order is the distinguishing technical characteristic that separates AIFF from WAV, which uses little-endian PCM (pcm_s16le). The source MP4 audio is almost always AAC at 128k–256k bitrate, which decodes to 16-bit PCM samples — making pcm_s16be a natural and lossless-in-the-remaining-sense target codec. If the source MP4 happens to contain MP3 or Opus audio tracks (less common but valid per the MP4 spec), those will also be decoded to PCM during this conversion. AIFF does not support multiple audio tracks, chapters, or subtitle tracks, all of which MP4 can carry — these are silently dropped. The -vn flag ensures the video stream is excluded, which prevents FFmpeg from raising an error about the lack of a video codec for the AIFF container. File sizes for AIFF are predictable: stereo 44.1 kHz 16-bit audio occupies approximately 10.1 MB per minute, compared to roughly 1 MB per minute for 128k AAC in an MP4.

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