Convert MKV to AIFC — Free Online Tool
Convert MKV video files to AIFC audio format, extracting the audio stream and encoding it as big-endian PCM (pcm_s16be) — the standard codec for Apple's compressed AIFF container. This is ideal for pulling broadcast-quality, uncompressed audio from MKV sources into a format compatible with professional macOS and Pro Tools workflows.
to
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
During this conversion, FFmpeg discards all video, subtitle, and chapter data from the MKV container — none of it is compatible with AIFC, which is a pure audio format. The audio stream (which in an MKV may be AAC, MP3, Opus, Vorbis, or FLAC) is decoded to raw PCM and then re-encoded as 16-bit big-endian PCM (pcm_s16be), the default codec for AIFC. Big-endian byte ordering is a defining characteristic of AIFC inherited from its origins on Motorola-based Apple hardware. If the source MKV contains multiple audio tracks, FFmpeg will select the first one by default. The result is an AIFC file with linear PCM audio — perceptually lossless relative to CD-quality sources, and immediately usable in Logic Pro, GarageBand, Pro Tools, and other professional Apple-ecosystem tools.
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.mkv
|
Specifies the input file as an MKV (Matroska) container, which may contain video, multiple audio tracks, subtitles, and chapters. FFmpeg reads and demuxes all streams from this container before applying the output options. |
-c:a pcm_s16be
|
Sets the audio codec to signed 16-bit big-endian PCM, which is the standard uncompressed audio format for AIFC files. This re-encodes whatever audio codec was in the MKV (AAC, FLAC, Opus, etc.) into raw PCM conforming to the AIFC/AIFF big-endian byte order specification. |
-b:a 128k
|
Nominally sets a target audio bitrate of 128 kbps, though this flag has no practical effect on pcm_s16be output because uncompressed PCM has a fixed bitrate determined by sample rate and bit depth rather than a configurable bitrate target. |
output.aifc
|
Specifies the output filename with the .aifc extension, which tells FFmpeg to wrap the encoded PCM audio in an AIFC container. The AIFC format is Apple's extension of AIFF and is natively supported by macOS, Logic Pro, Pro Tools, and other professional audio applications. |
Common Use Cases
- Extracting a clean, uncompressed audio track from an MKV film or documentary for use as a reference audio file in a professional macOS post-production environment
- Pulling the audio from an MKV screen recording or lecture capture and converting it to AIFC for import into Logic Pro or GarageBand for editing and cleanup
- Converting MKV-packaged music videos or concert recordings to AIFC so the audio can be mastered or mixed in a Pro Tools session that requires AIFF-family input files
- Archiving the audio component of an MKV file as big-endian PCM in AIFC format for long-term preservation, especially when the source audio is already lossless FLAC and re-encoding to a lossy format is undesirable
- Preparing audio from MKV game recordings or video essays for use in broadcast workflows on macOS systems where AIFC is a natively supported interchange format
- Stripping audio from an MKV file received from a Linux or Windows editor and delivering it to an Apple-based audio engineer in the AIFC format their software expects
Frequently Asked Questions
It depends on what audio codec is in your MKV file. If the source audio is already lossy — AAC, MP3, Opus, or Vorbis — those artifacts are already baked in, and the re-encoding to pcm_s16be PCM will not make things worse beyond negligible floating-point rounding. If the source audio is lossless FLAC, the conversion to 16-bit PCM is essentially transparent at normal listening levels, though you technically discard any bit depth above 16 bits. In no case does this conversion add new audible compression artifacts, because PCM is an uncompressed format.
AIFC inherits its byte ordering from the original AIFF specification, which was designed by Apple in the era of Motorola 68000 processors — a big-endian architecture. The pcm_s16be codec (signed 16-bit big-endian PCM) is the canonical audio format for AIFC files and is what professional Apple software like Logic Pro and Pro Tools expects. Little-endian PCM (like pcm_s16le, used in WAV files) would not conform to the AIFC specification, even though modern CPUs are typically little-endian internally.
All of them are dropped. AIFC is a pure audio container with no support for subtitles, chapters, or multiple tracks — only a single audio stream can exist in an AIFC file. FFmpeg will automatically select the first audio track from the MKV and convert only that. If you need to extract a specific audio track other than the first one, you would need to add a stream selector flag like -map 0:a:1 to the FFmpeg command before the output filename.
Replace pcm_s16be with another supported AIFC codec to change the bit depth or numeric format. For 24-bit PCM (common in professional audio), use -c:a pcm_s24be. For 32-bit PCM, use -c:a pcm_s32be. For 32-bit or 64-bit floating-point PCM, use -c:a pcm_f32be or -c:a pcm_f64be respectively. Telephony applications may call for G.711 codecs: -c:a pcm_alaw or -c:a pcm_mulaw. Higher bit depths preserve more dynamic range and are preferred when the AIFC file will go through further processing in a DAW.
Yes. On Linux or macOS you can use a shell loop: for f in *.mkv; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.mkv}.aifc"; done. On Windows Command Prompt, use: for %f in (*.mkv) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc". The browser-based tool on this page processes one file at a time, so the FFmpeg command is especially valuable for batch jobs involving many MKV files or files larger than 1GB.
In practice, PCM codecs like pcm_s16be have a fixed, mathematically determined bitrate based on sample rate and bit depth — for standard 44.1 kHz stereo 16-bit audio that works out to roughly 1411 kbps, far above 128k. FFmpeg accepts the -b:a 128k flag without error but it has no real effect on uncompressed PCM output; the actual bitrate is dictated entirely by the codec parameters. The flag would only meaningfully affect output if you were using a compressed AIFC codec like pcm_alaw or pcm_mulaw.
Technical Notes
AIFC (Audio Interchange File Format Compressed) is a superset of Apple's AIFF format that supports both uncompressed PCM and a range of compressed codecs, though in practice modern usage almost always means PCM in one of several bit depths. This conversion strips the MKV container entirely — including its video streams, embedded subtitle tracks (ASS, SRT, PGS, etc.), chapter markers, and any secondary audio tracks — because AIFC has no mechanism to carry any of that data. Only the first audio stream is extracted. The output uses pcm_s16be, which is 16-bit signed big-endian PCM; this is byte-for-byte compatible with standard AIFF and is natively readable by macOS Core Audio, Logic Pro, GarageBand, Pro Tools, and most professional audio software. If your MKV source audio was encoded at a higher sample rate than 44.1 kHz (e.g., 48 kHz from a video production pipeline), FFmpeg preserves that sample rate in the AIFC output — no resampling occurs unless you explicitly request it with -ar. Metadata like track title or artist embedded in the MKV is generally not preserved in the AIFC output, as AIFC has limited metadata support compared to containers like MKV or MP4. File sizes will be substantially larger than the source MKV in most cases, since you are replacing a compressed audio codec (AAC, Opus, etc.) with uncompressed PCM.