Convert MP4 to AIF — Free Online Tool
Extract and convert audio from MP4 video files into AIF format — Apple's lossless uncompressed PCM audio container. The audio stream is decoded from its compressed source codec (typically AAC or MP3) and re-encoded as 16-bit big-endian PCM, giving you a high-fidelity, uncompressed audio file ready for use in Logic Pro, GarageBand, or professional Mac audio workflows.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
MP4 files store audio in a compressed codec — most commonly AAC, though MP3 and Opus are also possible. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio into raw PCM samples, then re-encodes those samples as pcm_s16be (signed 16-bit, big-endian PCM) wrapped in Apple's AIFF container. Because AIF is uncompressed, the output file will be significantly larger than the source audio track — a 4-minute AAC track at 128k might expand from ~4MB to ~40MB as lossless 16-bit AIF. The conversion involves one decode step (AAC → raw PCM) and one encode step (raw PCM → pcm_s16be), so there is a single generation of lossy-to-lossless transcoding rather than a lossless copy.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles the decoding of the MP4 container and re-encoding into AIF format. In the browser version, this runs as a WebAssembly (FFmpeg.wasm) instance with no server involvement. |
-i input.mp4
|
Specifies the input file — an MP4 container that may hold a video stream, one or more audio tracks (typically AAC-encoded), and optionally subtitles or chapters. FFmpeg reads and demuxes this container to access the individual streams. |
-c:a pcm_s16be
|
Sets the audio codec to pcm_s16be — signed 16-bit big-endian PCM — which is the standard uncompressed audio encoding used inside AIF files. This decodes the compressed AAC audio from the MP4 and re-encodes it as raw 16-bit PCM samples in Apple's preferred byte order, making the output natively compatible with macOS audio applications. |
output.aif
|
Specifies the output filename with the .aif extension, which tells FFmpeg to wrap the pcm_s16be audio stream in Apple's AIFF container format. The video stream from the MP4 is automatically excluded because AIF is an audio-only container. |
Common Use Cases
- Extracting a musical performance or song from an MP4 video to import into Logic Pro or GarageBand as an editable audio asset
- Pulling dialogue or voiceover audio from an MP4 interview or documentary for post-production editing in a Mac-based DAW that prefers AIFF
- Converting an MP4 video soundtrack into AIF for mastering engineers who require uncompressed source files before mixing
- Archiving the audio component of an MP4 video in a lossless format for long-term preservation without re-encoding artifacts
- Preparing audio from an MP4 screen recording or tutorial video for use as samples or stems in music production software on macOS
- Extracting audio from MP4 files delivered by a client to produce uncompressed AIFF versions required by broadcast or post-production pipelines
Frequently Asked Questions
Not in the strictest sense. The AIF output is an uncompressed, lossless representation of the audio data decoded from the AAC stream, but AAC itself is a lossy codec — meaning some audio information was already discarded when the MP4 was originally created. The conversion faithfully preserves every bit of audio present in the MP4 source, but it cannot recover what AAC compression previously removed. Think of it as a lossless snapshot of a lossy original: no further quality is lost during this conversion, but the ceiling is the quality of the source AAC track.
MP4 typically stores audio as AAC, which achieves compression ratios of 10:1 or higher. AIF using pcm_s16be is completely uncompressed — every audio sample is stored as a raw 16-bit integer. A 5-minute AAC track encoded at 128 kbps occupies roughly 5MB, while the same audio as 16-bit stereo AIF at 44.1 kHz will be approximately 50MB. This size increase is expected and is the nature of converting from a compressed format to an uncompressed one.
Yes — regardless of the original AAC bitrate, the decoder fully decodes the compressed audio and the pcm_s16be encoder captures the resulting 16-bit PCM signal without further degradation. A 256k or 320k AAC source will produce a noticeably higher quality AIF than a 64k source because the AAC decoder has more data to work with. The FFmpeg command does not impose any bitrate ceiling during this specific conversion — audio quality in the output is bounded only by the source.
They are all discarded. AIF is a pure audio-only container — it has no support for video streams, subtitle tracks, chapter markers, or multiple audio tracks. FFmpeg automatically extracts only the first audio stream from the MP4 and ignores everything else. If your MP4 contains multiple audio tracks (e.g., different language dubs), only the first track will appear in the output. There is no way to embed the other MP4 features into AIF.
Replace pcm_s16be in the command with another supported AIF codec to change the bit depth. For 24-bit output use pcm_s24be, for 32-bit integer use pcm_s32be, and for 32-bit or 64-bit floating-point use pcm_f32be or pcm_f64be respectively. For example: ffmpeg -i input.mp4 -c:a pcm_s24be output.aif. If your source audio was originally recorded or mastered at 24-bit, using pcm_s24be will preserve that additional dynamic range rather than truncating to 16-bit.
Yes, using a shell loop. On macOS or Linux, run: for f in *.mp4; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mp4}.aif"; done. On Windows Command Prompt, use: for %f in (*.mp4) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aif". This processes every MP4 in the current directory and writes a matching AIF file for each. The in-browser tool handles one file at a time, making the FFmpeg command especially useful for bulk conversion of large collections.
Technical Notes
AIF (Audio Interchange File Format) stores audio as big-endian PCM, which reflects its Apple/Motorola heritage — this distinguishes it from WAV, which uses little-endian PCM. The default codec in this conversion is pcm_s16be, producing 16-bit signed big-endian samples compatible with virtually all Apple software including Logic Pro, GarageBand, Final Cut Pro, and Core Audio on macOS. The sample rate of the output matches the source MP4 audio stream — no resampling occurs unless explicitly requested with the -ar flag. If the source MP4 contains a 44.1 kHz or 48 kHz AAC track, the AIF output will reflect that same rate. AIF does not support album art, lyrics, chapter markers, or embedded subtitles, so any such metadata present in the MP4 container will be silently dropped. ID3-style tags are also not supported natively by AIF in the same way as MP3 or AAC — basic metadata like track name may or may not round-trip cleanly depending on the downstream application. For professional audio archiving at higher bit depths, consider specifying pcm_s24be, which is commonly preferred in studio workflows and is supported by all major AIF-compatible DAWs.