Extract Audio from MP4 to AIFC — Free Online Tool
Extract audio from MP4 video files and save it as AIFC, using the 16-bit big-endian PCM codec (pcm_s16be) for high-fidelity, uncompressed audio output. AIFC is the professional-grade extension of Apple's AIFF format, ideal for archival work and compatibility with pro audio software on macOS.
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FFmpeg Command
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DownloadHow It Works
This conversion strips the video stream entirely from the MP4 container and re-encodes the audio — typically AAC or MP3 inside the MP4 — into uncompressed 16-bit big-endian PCM audio wrapped in an AIFC container. Because AAC (the most common MP4 audio codec) is a lossy format and pcm_s16be is uncompressed, the conversion involves decoding the compressed AAC audio to raw PCM samples and then writing them into the AIFC file. This means the output will be significantly larger than the input audio stream, but there is no additional quality loss beyond what was already introduced when the MP4 was originally encoded. The video stream is discarded entirely using the -vn flag, so only the audio data is written to the output file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all media demuxing, decoding, encoding, and muxing in this conversion pipeline. |
-i input.mp4
|
Specifies the input MP4 file. FFmpeg reads the file, identifies the container as MPEG-4, and detects the contained streams — typically an H.264 video track and an AAC audio track. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the video stream in the MP4. Since AIFC is an audio-only format, this ensures no video data is passed to the muxer, producing a clean audio-only output. |
-c:a pcm_s16be
|
Sets the audio encoder to pcm_s16be — signed 16-bit big-endian PCM — which decodes the compressed AAC audio from the MP4 and writes it as uncompressed raw samples in the byte order native to the AIFC format. |
-b:a 128k
|
Specifies a 128k audio bitrate target. For uncompressed PCM codecs like pcm_s16be, this parameter has minimal practical effect since PCM bitrate is determined by sample rate and bit depth rather than a compression target; the actual output bitrate will reflect the source audio's sample rate and channel count. |
output.aifc
|
Defines the output filename and extension. FFmpeg uses the .aifc extension to select the AIFF-C muxer, which wraps the pcm_s16be audio data in a valid AIFC container readable by macOS audio applications and professional DAWs. |
Common Use Cases
- Extracting the audio track from an MP4 interview or lecture recording to import into Logic Pro or Pro Tools as a lossless AIFC file for professional editing
- Archiving the audio from MP4 video masters in an uncompressed AIFC format to ensure no further generation loss in a post-production pipeline
- Pulling audio from an MP4 film clip to use as a sound effect or dialogue sample in a macOS-based DAW that natively reads AIFC files
- Converting MP4 audio to AIFC for analysis in scientific or broadcast tools that require raw PCM audio in a big-endian AIFF-compatible container
- Preparing clean, uncompressed audio from an MP4 source for further processing steps — such as noise reduction or mastering — where a lossless intermediate format is preferred over keeping the lossy AAC stream
Frequently Asked Questions
No — and this is an important distinction. The audio inside most MP4 files is encoded with AAC, which is a lossy codec. When we decode that AAC audio and write it to an uncompressed AIFC file, you get a lossless representation of the already-decoded audio, but the quality ceiling is still set by the original AAC encoding. You will not lose any additional quality in this conversion, but you also cannot recover detail that AAC discarded when the MP4 was first created.
AAC, the default audio codec in MP4, achieves compression ratios of roughly 10:1 or more compared to uncompressed PCM. The AIFC output uses pcm_s16be, which stores every audio sample as a raw 16-bit value with no compression. A 10-minute MP4 with 128k AAC audio might produce an AIFC file that is 10–15 times larger in audio data alone. This is expected behavior for lossless PCM formats and is why AIFC is used as an archival or editing intermediate rather than a delivery format.
pcm_s16be stands for PCM, signed 16-bit, big-endian — meaning audio samples are stored as signed 16-bit integers in big-endian byte order, which is the native byte order for AIFF and AIFC on Apple platforms. 16-bit PCM matches the resolution of CD audio and is sufficient for most professional editing and archival tasks. If your source MP4 has high-dynamic-range audio and you need 24-bit fidelity, you could modify the FFmpeg command to use pcm_s24be instead.
Most common ID3-style metadata embedded in MP4 files (such as title, artist, or album) will not transfer cleanly to AIFC, because the two formats use different metadata schemes. MP4 uses iTunes-style atom-based metadata, while AIFC uses AIFF chunk-based metadata. FFmpeg will attempt to map common fields, but results vary, and specialized tags are often dropped. If metadata preservation is critical, you should verify the output file in a tag editor after conversion.
Yes. AIFC supports several PCM variants beyond the default pcm_s16be. You can substitute -c:a pcm_s24be for 24-bit audio, pcm_s32be for 32-bit integer, or pcm_f32be and pcm_f64be for 32-bit and 64-bit floating-point formats respectively. AIFC also supports the lossy codecs pcm_alaw and pcm_mulaw for telephony-grade audio. For most professional audio work, pcm_s24be is the recommended upgrade from the default 16-bit setting if your source material warrants it.
On macOS or Linux, you can loop over all MP4 files in a directory with: for f in *.mp4; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mp4}.aifc"; done. On Windows Command Prompt, use: for %f in (*.mp4) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aifc". This applies the same extraction and encoding settings to every file in the folder, outputting one AIFC file per MP4.
Technical Notes
AIFC (Audio Interchange File Format Compressed) is a superset of Apple's original AIFF format, adding support for compressed audio codecs alongside traditional uncompressed PCM. The format uses big-endian byte ordering, which is native to the Motorola 68000 and PowerPC architectures that Apple historically used, and is fully supported in modern macOS audio toolchains including Logic Pro, GarageBand, and Core Audio. The default codec used here — pcm_s16be — produces standard CD-quality, 16-bit stereo audio with no lossy compression stage in the output. One important limitation is that AIFC does not support multiple audio tracks, chapter markers, or subtitle streams, all of which MP4 can carry; only the first (or default) audio track from the MP4 will be extracted. If your MP4 contains multiple audio tracks and you need a specific one, you can add -map 0:a:1 (or the appropriate track index) to the FFmpeg command before the output filename. Also note that the -b:a bitrate flag has limited effect on uncompressed PCM codecs like pcm_s16be — bitrate is effectively determined by the sample rate and bit depth, not a compression target.