Extract Audio from MPG to AIFC — Free Online Tool
Extract audio from MPG video files and save it as AIFC, converting the MP2 or MPEG audio stream into big-endian PCM audio using the pcm_s16be codec. Ideal for professionals who need uncompressed, high-fidelity audio from legacy MPEG-1/2 broadcast or DVD source material.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your MPG file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
Settings
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
MPG files typically carry audio encoded as MP2 (MPEG-1 Audio Layer II), the standard codec for VCD and broadcast MPEG streams, sometimes alongside MP3 or AAC. This tool strips the video stream entirely and decodes the compressed MPG audio, then re-encodes it into AIFC using the pcm_s16be codec — 16-bit signed big-endian PCM. AIFC is an extension of Apple's AIFF format and natively stores audio in big-endian byte order, making pcm_s16be the natural fit. The result is an uncompressed audio file where the original lossy MP2 compression artifacts are preserved but no further quality is lost in the conversion step itself — the decoded audio is written directly to PCM without any additional lossy encoding.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that powers this conversion. In the browser, this runs via FFmpeg.wasm compiled to WebAssembly. |
-i input.mpg
|
Specifies the input MPG file. FFmpeg reads the MPEG-1 or MPEG-2 program stream and identifies the audio track (typically MP2) and video track for demuxing. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the mpeg1video or mpeg2video stream in the MPG file. This is what makes the operation an audio extraction rather than a full conversion. |
-c:a pcm_s16be
|
Encodes the decoded MPG audio into 16-bit signed big-endian PCM, the native and standard uncompressed codec for AIFC files. The MP2 audio is fully decoded first, then written as raw PCM — no additional lossy compression is applied. |
-b:a 128k
|
Sets a target audio bitrate of 128k, though this flag has no practical effect on uncompressed PCM output — pcm_s16be bitrate is fixed by sample rate and channel count, not a variable bitrate target. It is included for command consistency but does not alter the output. |
output.aifc
|
Defines the output filename and signals to FFmpeg that the file should be wrapped in the AIFC container format, which supports big-endian PCM audio and is natively compatible with Apple and professional audio applications. |
Common Use Cases
- Extracting dialogue or narration from a VCD or DVD-sourced MPG file for use in a professional audio editing session in Logic Pro or Pro Tools, which natively support AIFC
- Recovering audio from archival MPEG-2 broadcast recordings for digitization workflows that require uncompressed PCM masters in AIFC format
- Pulling the MP2 audio track from an MPEG-1 video file to analyze waveforms or perform noise reduction in a DAW without further lossy re-encoding
- Extracting music or soundtrack audio from MPEG video files produced by legacy broadcast equipment or VCD authoring tools for remix or restoration projects
- Converting the audio track of an educational or training video stored in MPG format into a standalone AIFC file for use in Apple-ecosystem production pipelines
- Isolating audio from MPEG-2 broadcast captures for compliance review or transcription workflows that require an editable, uncompressed audio file
Frequently Asked Questions
No — the original MP2 audio in the MPG file is a lossy compressed format, and any compression artifacts it introduced are already baked into the signal. Converting to AIFC with pcm_s16be produces an uncompressed PCM file, but it preserves the decoded MP2 signal exactly as-is without adding further degradation. Think of it as locking in the current quality rather than improving it — AIFC simply provides a lossless container for what was already there.
AIFC inherits its byte order from the original AIFF specification, which was designed for Motorola and Apple hardware that uses big-endian byte ordering. The pcm_s16be codec (16-bit signed big-endian PCM) is the correct and native PCM format for AIFC files, ensuring full compatibility with tools and platforms that expect standard AIFC audio. Using little-endian PCM in an AIFC container would be non-standard and could cause playback issues in some applications.
FFmpeg preserves the sample rate and channel count from the source MPG audio stream. MPG files from VCD sources typically carry 44.1 kHz stereo MP2 audio, while MPEG-2 broadcast or DVD sources commonly use 48 kHz stereo or occasionally mono. The output AIFC file will mirror whatever sample rate and channel configuration was in the original MPG without any resampling or downmixing, unless you add explicit flags to the command.
Because the output is uncompressed PCM (pcm_s16be), the -b:a 128k bitrate flag in the command has no functional effect — PCM audio bitrate is determined entirely by sample rate, bit depth, and channel count, not a target bitrate setting. If you want higher bit depth, you can replace pcm_s16be with pcm_s24be or pcm_s32be in the command, for example: ffmpeg -i input.mpg -vn -c:a pcm_s24be output.aifc. This increases dynamic range headroom but also increases file size proportionally.
Significantly larger. MP2 audio at 192 kbps (a typical MPG bitrate) compresses audio roughly 6:1 compared to 16-bit PCM at 44.1 kHz stereo, which runs at about 1,411 kbps. So a one-hour MPG audio track that occupies roughly 86 MB as MP2 would expand to approximately 600 MB as uncompressed AIFC pcm_s16be. This is expected behavior when moving from a lossy compressed format to uncompressed PCM.
Yes. On Linux or macOS you can use a shell loop: for f in *.mpg; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.mpg}.aifc"; done. On Windows Command Prompt, use: for %f in (*.mpg) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aifc". This processes each MPG file in the current directory and outputs a matching AIFC file, which is especially useful when extracting audio from a collection of archival MPEG recordings.
Technical Notes
MPG audio streams are almost universally MP2 (MPEG-1 Audio Layer II) for VCD-spec content, though MPEG-2 program streams from broadcast or DVD sources may carry MP2 at 48 kHz or occasionally AAC. FFmpeg reliably demuxes all of these. The output AIFC format using pcm_s16be offers a 96 dB dynamic range ceiling — sufficient for most broadcast and archival use cases, though 24-bit PCM (pcm_s24be) would be preferable if the source material is high-quality broadcast content and maximum headroom is desired. AIFC files produced by this conversion are compatible with macOS's built-in audio tools, Logic Pro, Final Cut Pro, and most professional DAWs, making them well-suited for Apple-centric post-production pipelines. One limitation to be aware of: AIFC does not support embedded chapter markers or multiple audio tracks, but since MPG also lacks these features, nothing is lost in that regard. Metadata embedded in the MPG container (such as creation date or title tags) is not transferred to the AIFC output by this command, as AIFC has limited metadata support compared to formats like FLAC or WAV with INFO chunks.