Convert MP4 to AIFF — Free Online Tool
Extract and convert audio from an MP4 video file into a high-quality, uncompressed AIFF file using PCM 16-bit big-endian encoding. AIFF is Apple's native lossless audio container, making this conversion ideal for bringing video soundtracks into professional macOS audio workflows without any quality loss.
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FFmpeg Command
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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:
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DownloadHow It Works
MP4 files store audio in compressed formats such as AAC or MP3, which use lossy compression to reduce file size. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio track, then re-encodes it as raw PCM data (pcm_s16be — 16-bit signed, big-endian) wrapped in the AIFF container. This means the lossy audio from the MP4 is fully decompressed into uncompressed PCM, producing a lossless AIFF file. Importantly, because the source audio in the MP4 was originally lossy (typically AAC), the AIFF output will be uncompressed but not higher fidelity than the source — it will be a lossless representation of the decoded lossy signal, with no further generation loss if used in subsequent editing.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (ffmpeg.wasm) — your file never leaves your device. |
-i input.mp4
|
Specifies the input file — an MP4 container that may hold H.264 or H.265 video alongside compressed audio (typically AAC). FFmpeg reads all available streams from this file before applying the output settings. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, which is the standard uncompressed audio encoding used inside AIFF files. This decodes whatever compressed audio format (such as AAC) was in the MP4 and writes it as raw, uncompressed PCM samples in the big-endian byte order required by the AIFF specification. |
output.aiff
|
Defines the output filename and format. The .aiff extension tells FFmpeg to wrap the PCM audio stream in an AIFF container — Apple's lossless audio format designed for high-quality storage and professional use on macOS. No video or subtitle streams are written since AIFF does not support them. |
Common Use Cases
- Importing a music video or live performance recording's audio track into Logic Pro or GarageBand, which prefer AIFF for native project assets on macOS
- Extracting dialogue or narration from an MP4 screen recording or lecture video to use as an uncompressed source clip in a professional audio post-production session
- Archiving the audio from MP4 video files in an uncompressed format before the original video source is deleted, preserving the decoded audio without further compression artifacts
- Pulling a film score or sound design audio track from an MP4 for use in a sample library, where AIFF is the expected format for sample players like Kontakt or Logic's EXS24
- Converting an MP4 podcast recording that was delivered as video into an AIFF file for mastering in a DAW before re-exporting as a compressed audio format for distribution
Frequently Asked Questions
Not in the purest sense — the AIFF will be an uncompressed, lossless representation of the decoded AAC audio, but it cannot recover information that was discarded during the original AAC encoding. Think of it as freezing the decoded signal: no additional quality is lost in this conversion, and no further compression artifacts will be introduced, but you won't regain fidelity lost when the AAC was first created. If the source MP4 audio was encoded at a high AAC bitrate (192k or above), the resulting AIFF will sound excellent for most professional purposes.
AIFF stores audio as raw, uncompressed PCM data — every sample is stored at full resolution with no compression whatsoever. A typical AAC audio track in an MP4 at 128k bitrate might occupy around 1MB per minute, while a 16-bit stereo AIFF at 44.1kHz consumes roughly 10MB per minute. This size increase is completely expected and is the nature of uncompressed audio. The video stream is also removed, but the audio decompression is the dominant factor in the size change.
AIFF supports a limited set of metadata via ID3 tags embedded in the file, but FFmpeg does not automatically map all MP4 metadata fields (such as iTunes-style atoms) to AIFF during this conversion. Common tags like title and artist may transfer, but artwork and extended metadata fields are likely to be dropped. If metadata preservation is critical, you should verify the output in your target application and consider adding tags manually using a tool like Kid3 or iTunes after conversion.
pcm_s16be stands for PCM signed 16-bit big-endian — this is standard CD-quality audio depth, the same used on audio CDs, and is compatible with virtually all applications that open AIFF files. For professional audio workflows where the source material warrants it, you may prefer 24-bit (pcm_s24be) or 32-bit (pcm_s32be) output for greater dynamic range headroom during editing. However, since the source is a compressed AAC or MP3 track from an MP4, using 16-bit is typically sufficient — the higher bit depths won't recover lost information but do provide more headroom if you plan to apply significant gain or processing in a DAW.
Replace pcm_s16be with your desired codec in the command. For 24-bit output use: ffmpeg -i input.mp4 -c:a pcm_s24be output.aiff. For 32-bit integer use pcm_s32be, and for 32-bit float use pcm_f32be. The 24-bit option (pcm_s24be) is the most common choice in professional audio production, matching the standard bit depth of modern recording sessions and providing a larger dynamic range than 16-bit.
Yes — on macOS or Linux you can run a shell loop: for f in *.mp4; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mp4}.aiff"; done. On Windows Command Prompt use: for %f in (*.mp4) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff". This will process every MP4 in the current directory, outputting an AIFF file for each. This is particularly useful for archiving a folder of video files as uncompressed audio assets.
Technical Notes
This conversion involves audio-only output — the video, subtitle, and chapter streams present in the MP4 are all dropped, as AIFF is a pure audio container with no support for these stream types. The default codec used is pcm_s16be (16-bit signed big-endian PCM), which matches the AIFF specification and is natively readable on macOS without any additional codecs. AIFF uses big-endian byte ordering, inherited from its origins on Motorola 68k-based Macs, which distinguishes it from WAV (which uses little-endian PCM). If your MP4 contains multiple audio tracks, FFmpeg will by default select the first (or best-ranked) audio stream; to extract a specific track, add -map 0:a:1 (for the second audio track) before the output filename. The sample rate of the output AIFF will match whatever is present in the MP4 source — typically 44100Hz or 48000Hz — so no sample rate conversion occurs unless explicitly requested with the -ar flag. One known limitation is that if the MP4 contains a Dolby AC-3 or surround audio track, the PCM output will decode all channels, which may result in a multi-channel AIFF; not all applications handle multi-channel AIFF gracefully, so you may want to add -ac 2 to downmix to stereo if needed.