Extract Audio from MKV to AIFC — Free Online Tool
Extract audio from MKV video files and save it as AIFC — Apple's professional compressed audio container — using PCM 16-bit big-endian encoding for uncompressed, broadcast-quality audio. Ideal for audio engineers and Mac users who need lossless-quality audio extracted from Matroska video files in a format compatible with professional Apple and pro-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
MKV files commonly carry audio encoded in lossy formats like AAC, MP3, Opus, or Vorbis, or losslessly in FLAC. During this conversion, the video stream is completely discarded (no video is written to the output), and the audio stream is decoded from whatever codec it was stored in inside the MKV container, then re-encoded into PCM signed 16-bit big-endian audio — the default codec for AIFC. This is a full audio transcode: the raw audio samples are decoded to PCM and then written into the AIFC container with big-endian byte ordering, as required by Apple's AIFF/AIFC specification. If your MKV contains multiple audio tracks, only the default (first) track is extracted unless you specify otherwise. Subtitle tracks, chapter markers, and metadata from the MKV are not carried over, as AIFC does not support those features.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop command line. |
-i input.mkv
|
Specifies the input file — in this case an MKV (Matroska) container, which may contain video, one or more audio tracks (AAC, MP3, FLAC, Opus, etc.), subtitles, and chapter data. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore all video streams from the MKV and write only audio to the AIFC file. Without this flag, FFmpeg would attempt to include video, which AIFC cannot contain. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, the standard uncompressed audio format required by the AIFC specification. This fully decodes whatever compressed audio (AAC, Vorbis, etc.) was in the MKV into raw PCM samples stored in big-endian byte order. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For PCM codecs like pcm_s16be, bitrate is determined by sample rate and bit depth rather than this parameter, so this flag has minimal practical effect on uncompressed output but is included for explicit control if switching to a compressed AIFC codec. |
output.aifc
|
The output filename with the .aifc extension, which tells FFmpeg to wrap the decoded PCM audio in an AIFC container — Apple's extended AIFF format that supports both compressed and uncompressed audio data. |
Common Use Cases
- Extract a music or dialogue audio track from an MKV film or TV episode for editing in Logic Pro, GarageBand, or Pro Tools on macOS, where AIFC is natively supported.
- Convert a FLAC or AAC audio track embedded in an MKV file to uncompressed PCM AIFC for archival or mastering purposes where a fully decoded, editor-ready audio file is required.
- Pull audio from an MKV screen recording or lecture capture into AIFC format for transcription or audio post-production on Apple hardware.
- Prepare audio stems from MKV video deliverables for broadcast workflows that require PCM audio in big-endian AIFC format, common in legacy Mac-based post-production pipelines.
- Extract and decode a compressed audio track (e.g., Opus or Vorbis) from an MKV into a lossless PCM representation for forensic audio analysis or quality comparison.
- Convert MKV audio to AIFC as an intermediate step before further encoding to Apple Lossless (ALAC) or other formats within a macOS audio pipeline.
Frequently Asked Questions
It depends on what audio codec is stored in your MKV. If the source MKV contains lossy audio — such as AAC, MP3, Opus, or Vorbis — decoding it to PCM AIFC will not recover the quality lost during the original lossy encoding, though the resulting AIFC will be an uncompressed representation of that audio. If the MKV contains FLAC audio, the conversion to 16-bit PCM AIFC is effectively lossless, assuming the original was also 16-bit. In no case does this process add new compression artifacts, but you cannot undo pre-existing lossy compression.
AIFC supports both uncompressed PCM and several compressed formats, but pcm_s16be (signed 16-bit big-endian PCM) is the most universally compatible AIFC codec and is the format's standard baseline. Big-endian byte ordering is a requirement of the AIFF/AIFC specification, which originated on Motorola 68k and PowerPC Apple hardware. Using PCM ensures maximum compatibility with any software that can read AIFC files, including older Pro Tools versions and Apple's own audio frameworks.
Yes, but you'll need to modify the FFmpeg command. MKV supports multiple audio tracks, and by default FFmpeg extracts the first (default) audio stream. To extract a specific track, add the flag '-map 0:a:1' (for the second audio track, zero-indexed) before the output filename in the command. You can inspect all available tracks in your MKV by running 'ffmpeg -i input.mkv' and reading the stream list it prints.
For PCM audio in AIFC, the concept of bitrate is less meaningful than for lossy codecs — quality is primarily determined by the codec (bit depth) and the source sample rate. To use 24-bit PCM instead of 16-bit, replace '-c:a pcm_s16be' with '-c:a pcm_s24be' in the command. To preserve the original sample rate of the MKV audio rather than letting FFmpeg default, add '-ar 48000' (or whatever rate applies) before the output. The '-b:a 128k' flag has limited effect on PCM streams but is included for compatibility.
No. AIFC is a pure audio container and does not support chapters, subtitles, or embedded metadata beyond basic audio tags. All of those data streams from the MKV are silently dropped during conversion. If preserving chapter or subtitle data matters, you should retain the original MKV file alongside the extracted AIFC.
Yes. On Linux or macOS, you can run a simple shell loop: 'for f in *.mkv; do ffmpeg -i "$f" -vn -c:a pcm_s16be -b:a 128k "${f%.mkv}.aifc"; done'. On Windows (PowerShell), use: 'Get-ChildItem *.mkv | ForEach-Object { ffmpeg -i $_.FullName -vn -c:a pcm_s16be -b:a 128k ($_.BaseName + ".aifc") }'. This is especially useful for the desktop FFmpeg command shown on this page, since the browser tool processes one file at a time and has a 1GB file size limit.
Technical Notes
AIFC (Audio Interchange File Format Compressed) uses big-endian byte ordering inherited from Apple's original AIFF specification, which means the output file is not directly interchangeable with WAV (which uses little-endian PCM). The pcm_s16be codec produces 16-bit signed samples at whatever sample rate the source MKV audio carries — commonly 44.1 kHz for music or 48 kHz for video-originated content. FFmpeg will preserve the original sample rate unless you explicitly resample. If your MKV source audio was encoded at a higher bit depth (e.g., 24-bit FLAC), converting to pcm_s16be will quantize down to 16-bit, which technically reduces dynamic range from ~144 dB to ~96 dB — use pcm_s24be if preserving full depth is important. AIFC does not carry MKV metadata fields like title, language tags, or track names, so those are lost in conversion. The resulting AIFC files will be significantly larger than the source MKV audio if the original was lossy-compressed: a 10-minute AAC track at 128 kbps (~10 MB) becomes roughly 100 MB as uncompressed 16-bit/48 kHz PCM AIFC. AIFC is natively readable by macOS Core Audio, QuickTime, Logic Pro, and most professional DAWs, but has limited support outside the Apple and professional audio ecosystem compared to WAV.