MyMediaTools

Convert MP4 to AIFC — Free Online Tool

Convert MP4 video files to AIFC audio format, extracting the audio stream and re-encoding it to 16-bit big-endian PCM using the pcm_s16be codec. AIFC is Apple's compressed Audio Interchange File Format, widely used in professional audio workflows on macOS and compatible with legacy Apple hardware and pro audio software.

FFmpeg Command

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Estimated output:

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How It Works

During this conversion, FFmpeg discards the video stream entirely and extracts only the audio from the MP4 container. The source audio — typically AAC or MP3 encoded inside the MP4 — is decoded and then re-encoded to 16-bit signed big-endian PCM (pcm_s16be) and wrapped in an AIFC container. This is a full transcode of the audio: the compressed lossy audio from the MP4 is decoded to raw PCM and then written into the AIFC file. Because the output is uncompressed PCM, the resulting file will be significantly larger than the MP4's audio track, but it will be free of any lossy compression artifacts introduced after this step. The video, subtitle tracks, chapter markers, and additional audio tracks present in the MP4 are all dropped, as AIFC supports only a single audio stream with no video or metadata of that kind.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg program. In the browser-based tool, this runs via FFmpeg.wasm compiled to WebAssembly, executing entirely within your browser with no server involvement.
-i input.mp4 Specifies the input file — an MP4 container that may contain video, audio (typically AAC or MP3), subtitle tracks, and chapter markers. FFmpeg reads all streams but only the audio will be used in the output.
-c:a pcm_s16be Sets the audio codec to pcm_s16be — 16-bit signed big-endian PCM — which is the standard uncompressed audio encoding used in AIFC files and the most broadly compatible choice for Apple and professional audio software.
-b:a 128k Specifies an audio bitrate of 128 kbps. For the pcm_s16be codec this flag has no meaningful effect, since uncompressed PCM audio has a fixed bitrate determined by its sample rate and channel count rather than an encoder setting; it is included for consistency with the tool's interface but can safely be omitted when running the command locally.
output.aifc Defines the output filename with the .aifc extension, which signals to FFmpeg to wrap the encoded audio in an AIFC container (Apple's compressed Audio Interchange File Format) using its big-endian chunk structure.

Common Use Cases

  • Importing audio from an MP4 screen recording or video interview into Logic Pro or Pro Tools, which have historically preferred AIFF/AIFC files for session audio on macOS
  • Archiving the audio portion of an MP4 video to an uncompressed PCM format for long-term preservation, avoiding codec dependency on AAC or MP3 decoders
  • Delivering audio extracted from an MP4 to a client or broadcaster whose ingest pipeline requires AIFC files with big-endian PCM, common in older Apple-centric post-production environments
  • Stripping the audio from an MP4 promotional video to produce a clean, uncompressed audio asset for use in a radio spot or podcast intro without re-compressing to a lossy format
  • Extracting a music track embedded in an MP4 music video for use as a sample in Apple GarageBand or Soundtrack Pro, which natively read AIFC files
  • Converting MP4 audio to AIFC for import into legacy digital audio workstations or hardware samplers that read AIFF/AIFC but do not support AAC-wrapped audio

Frequently Asked Questions

No — and this is an important distinction. The audio inside most MP4 files is already encoded in a lossy format like AAC or MP3. When FFmpeg decodes that audio and re-encodes it to PCM in the AIFC file, the original lossy compression artifacts are baked into the uncompressed output. The AIFC file will be lossless from this point forward, meaning no further quality will be lost in subsequent edits, but the conversion cannot recover detail that was discarded when the MP4 was originally created. Think of it as 'lossless from now on', not 'lossless from the original source'.
MP4 files typically store audio as compressed AAC, which achieves very small file sizes by discarding audio data the human ear is less sensitive to. AIFC with pcm_s16be stores every audio sample as raw, uncompressed 16-bit data — approximately 5 MB per minute for a stereo 44.1 kHz stream, compared to roughly 1 MB per minute for 128 kbps AAC. There is no video stream in the output, but the uncompressed PCM audio alone will almost always be larger than the entire original MP4.
AIFC was designed as an extension of AIFF to support compressed audio codecs such as pcm_s16be, pcm_s24be, pcm_s32be, pcm_f32be, pcm_f64be, pcm_alaw, and pcm_mulaw. The pcm_s16be default (16-bit signed big-endian PCM) is chosen because it matches standard CD-quality audio resolution and has the broadest compatibility with Apple software and legacy hardware that reads AIFC files. If your target application requires higher bit depth, you can switch to pcm_s24be or pcm_s32be in the FFmpeg command.
No. AIFC is a purely audio-focused container and does not support video streams, subtitle tracks, chapter markers, or multiple audio tracks. All of these elements present in the MP4 will be silently dropped during conversion. Only the first (or default) audio stream from the MP4 is extracted and written to the AIFC output. If your MP4 contains multiple audio tracks, you would need to specify which track to extract using the '-map' flag in a custom FFmpeg command.
Replace pcm_s16be in the command with any AIFC-compatible PCM codec. For example, to produce 24-bit audio use 'ffmpeg -i input.mp4 -c:a pcm_s24be output.aifc', or for 32-bit float use 'ffmpeg -i input.mp4 -c:a pcm_f32be output.aifc'. Note that the '-b:a 128k' bitrate flag is largely irrelevant for uncompressed PCM codecs — the bitrate is determined by the sample rate and bit depth, not an encoder setting. You can safely omit '-b:a' when using pcm_s16be or other PCM variants.
Yes. On Linux or macOS, you can use a shell loop: 'for f in *.mp4; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.mp4}.aifc"; done'. On Windows Command Prompt, use: 'for %f in (*.mp4) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aifc"'. This applies the same pcm_s16be conversion to every MP4 in the current directory, producing a matching AIFC file for each one. The browser-based tool on this page processes one file at a time, so the FFmpeg command is the most practical route for batch jobs.

Technical Notes

The AIFC format stores audio in big-endian byte order, which reflects its origins on Motorola 68000-based Macintosh hardware where big-endian was the native byte order. Modern x86 and ARM systems are little-endian, so AIFC files are not natively the most efficient format for those platforms, but all major audio applications on macOS handle the byte-swap transparently. The pcm_s16be codec produces a 16-bit depth output regardless of the original MP4 audio's bit depth — if the source was encoded from a 24-bit master, some resolution is lost relative to what a pcm_s24be output would preserve. The '-b:a 128k' flag in the command has no practical effect on PCM codecs because uncompressed PCM has no configurable bitrate; the actual bitrate is entirely determined by the sample rate and channel count of the source audio. Metadata such as ID3 tags or iTunes metadata embedded in the MP4's audio stream may not transfer cleanly to AIFC, as AIFC uses its own chunk-based metadata scheme and FFmpeg's mapping between the two is limited. For professional audio workflows requiring metadata preservation, you may need to re-tag the AIFC file manually in an application like Sound Studio or fre:ac after conversion.

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