Extract Audio from MPEG to AIFC — Free Online Tool
Extract audio from MPEG video files and save it as AIFC, converting the MPEG audio stream (typically MP2 or MP3) into a big-endian PCM-encoded AIFC file using the pcm_s16be codec. Ideal for professional audio workflows that require uncompressed or losslessly archived audio from legacy broadcast and DVD-compatible MPEG sources.
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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:
Conversion Complete!
DownloadHow It Works
During this conversion, FFmpeg discards the MPEG video stream entirely and re-encodes the audio stream from its original compressed format (commonly MP2, the standard audio codec for MPEG-1 and MPEG-2 broadcast content) into 16-bit big-endian PCM audio (pcm_s16be) wrapped in an AIFC container. AIFC is Apple's compressed/extended AIFF format, but when using pcm_s16be the audio is stored as uncompressed linear PCM — meaning no lossy compression is applied at the output stage. However, because the source MPEG audio was already lossy (MP2 or similar), the output AIFC file will reflect the quality ceiling of the original compressed audio, not a true lossless capture. The result is a flat, uncompressed audio file with big-endian byte ordering, compatible with professional audio tools and legacy Apple hardware and software.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version of this tool, the same logic runs via FFmpeg.wasm (WebAssembly), which executes the identical processing pipeline client-side without any server upload. |
-i input.mpeg
|
Specifies the input MPEG file. FFmpeg will parse the MPEG program stream, identifying both the video stream (typically mpeg2video) and the audio stream (typically MP2) encapsulated within it. |
-vn
|
Disables video output entirely. Since this is an audio extraction tool, the MPEG video stream (mpeg1video or mpeg2video) is discarded and will not appear in the AIFC output, which is an audio-only format. |
-c:a pcm_s16be
|
Sets the output audio codec to 16-bit signed big-endian PCM, which is the native and standard uncompressed audio codec for the AIFC/AIFF container family. This decodes the lossy MP2 audio from the MPEG source into uncompressed linear PCM with big-endian byte ordering required by the AIFC format. |
-b:a 128k
|
Specifies a target audio bitrate of 128kbps. For the pcm_s16be codec used here, this flag does not constrain the actual output bitrate — uncompressed PCM bitrate is fixed by sample rate and bit depth — but it is included as a default parameter inherited from the tool's configuration. |
output.aifc
|
Defines the output filename and signals FFmpeg to write an AIFC container. The .aifc extension instructs FFmpeg to use the AIFF-C muxer, which wraps the pcm_s16be audio in a big-endian AIFC file compatible with Pro Tools, Logic Pro, and other professional Apple-ecosystem audio applications. |
Common Use Cases
- Extracting MP2 audio tracks from archived MPEG-2 broadcast recordings for ingestion into a professional DAW like Pro Tools or Logic Pro, which supports AIFC natively
- Pulling audio from DVD-compatible MPEG files to create uncompressed masters for archiving in a big-endian PCM format that preserves maximum fidelity from the source
- Converting legacy MPEG-1 video files from the 1990s or early 2000s into AIFC audio for use in classic Mac OS or early macOS audio editing environments
- Separating the audio commentary or narration track from an MPEG broadcast recording to use in a post-production pipeline that requires AIFF-family file formats
- Preparing audio extracted from MPEG streaming captures for quality analysis tools or audio forensics software that expects uncompressed big-endian PCM input
- Stripping and archiving the audio layer from MPEG-2 video masters before video re-encoding, preserving the audio in an uncompressed container to avoid any additional generation loss
Frequently Asked Questions
Not truly lossless, because MP2 (the default audio codec in MPEG-1 and MPEG-2 files) is itself a lossy format. FFmpeg will decode the MP2 audio and re-encode it into uncompressed pcm_s16be for the AIFC file, which means no further lossy compression is introduced at the output stage. However, the quality ceiling is set by the original MP2 encoding — any artifacts or frequency information lost in the original MPEG file cannot be recovered.
AIFC is derived from Apple's AIFF format, which historically used big-endian byte ordering to match Motorola 68000 and PowerPC processor architectures. The 'be' in pcm_s16be stands for big-endian, and it is the native and required byte order for AIFC/AIFF containers. Tools like Pro Tools, early versions of Logic, and other professional Apple-ecosystem audio applications expect this byte ordering, which is why pcm_s16be is the correct codec choice for this container.
The AIFC output will typically be significantly larger than the audio portion of the source MPEG file, because pcm_s16be is uncompressed PCM audio while the original MP2 audio in the MPEG file was compressed. At 128k bitrate reference, a one-hour MPEG file with a 192kbps MP2 track would produce an AIFC file of roughly 600MB for the audio alone, since uncompressed 16-bit stereo at 44.1kHz runs about 10MB per minute. The video stream is discarded, so you are not paying a size penalty for that.
MPEG files rarely carry rich metadata, and AIFC has limited metadata support compared to formats like FLAC or MP4. FFmpeg will attempt to copy any ID3 or container-level tags present in the MPEG source, but AIFC stores metadata in AIFF marker and annotation chunks, which may not map cleanly from MPEG metadata structures. For professional archiving workflows, it is advisable to add or verify metadata manually in the output AIFC file using a dedicated audio tag editor.
Yes. AIFC supports several other PCM codecs including pcm_s24be (24-bit), pcm_s32be (32-bit), pcm_f32be (32-bit float), and pcm_f64be (64-bit float), as well as lossy options like pcm_alaw and pcm_mulaw. For example, replacing '-c:a pcm_s16be' with '-c:a pcm_s24be' will produce a 24-bit AIFC file. However, since the source MPEG audio is MP2 (a lossy format with a quality ceiling well below 24-bit), increasing bit depth will increase file size without recovering any lost audio information.
On Linux or macOS, you can run a shell loop: 'for f in *.mpeg; do ffmpeg -i "$f" -vn -c:a pcm_s16be -b:a 128k "${f%.mpeg}.aifc"; done'. On Windows Command Prompt, use: 'for %f in (*.mpeg) do ffmpeg -i "%f" -vn -c:a pcm_s16be -b:a 128k "%~nf.aifc"'. This processes each MPEG file in the current directory and outputs a corresponding AIFC file, applying the same audio extraction and PCM encoding to all files in sequence.
Technical Notes
MPEG containers (both MPEG-1 and MPEG-2 program streams) multiplex video and audio into a single bitstream. The audio track is most commonly MP2 (MPEG-1 Audio Layer II), which is a lossy format widely used in broadcast television and DVD production. When FFmpeg extracts and decodes this stream for output as pcm_s16be AIFC, it performs a full decode-encode cycle: MP2 compressed audio → raw PCM → big-endian 16-bit PCM in AIFC container. The '-b:a 128k' flag in the command has no practical effect when using a fixed-rate uncompressed codec like pcm_s16be — the actual bitrate of the output is determined by sample rate, bit depth, and channel count (e.g., stereo 44.1kHz 16-bit = ~1411kbps). AIFC is not widely supported outside of Apple and professional audio ecosystems; if you need broad compatibility, consider WAV (pcm_s16le) instead. Also note that MPEG files do not support multiple audio tracks or subtitle streams, so there is no risk of accidentally omitting secondary audio content — what you see in the MPEG is the single audio stream that gets extracted.