Convert MPG to AIFC — Free Online Tool
Convert MPG video files to AIFC audio by extracting the MPEG-1/2 audio stream and re-encoding it as big-endian PCM audio (pcm_s16be), producing a lossless 16-bit stereo file compatible with professional Apple audio workflows. This is ideal when you need broadcast-quality audio extracted from legacy VCD or DVD-sourced MPEG footage.
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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
MPG files typically carry MP2 (MPEG Layer 2) or MP3 audio alongside an MPEG-1 or MPEG-2 video stream. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio (MP2/MP3) from the MPG container, then re-encodes it as PCM 16-bit big-endian audio and wraps it in an AIFC container. Because the source audio is lossy (MP2 or AAC at a fixed bitrate) and the output is lossless PCM, the conversion captures a faithful, uncompressed snapshot of whatever audio quality existed in the original MPG — it does not restore detail lost during the original MPEG compression. The resulting AIFC file is uncompressed and significantly larger than the audio portion of the source MPG.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles all media demuxing, decoding, encoding, and muxing steps in this conversion pipeline entirely within your browser via WebAssembly. |
-i input.mpg
|
Specifies the input MPG file, which FFmpeg will parse as an MPEG-1 or MPEG-2 program stream containing a compressed video track (mpeg1video or mpeg2video) and a compressed audio track (typically MP2). |
-c:a pcm_s16be
|
Decodes the MPG's compressed MP2 audio and re-encodes it as 16-bit signed big-endian PCM, which is the native uncompressed audio format expected by the AIFC container and Apple audio tools. |
-b:a 128k
|
Sets a target audio bitrate of 128 kbps; for the uncompressed pcm_s16be codec used here, this flag has no practical effect since PCM bitrate is fixed by sample rate and bit depth, but it is included for consistency with the tool's audio quality interface. |
output.aifc
|
Defines the output filename and instructs FFmpeg to mux the PCM audio stream into an AIFC container, the Apple-originated format that extends AIFF to support both uncompressed and compressed audio in a big-endian structure. |
Common Use Cases
- Extracting narration or dialogue from archived VCD or DVD-sourced MPG footage for use in a professional Apple Logic Pro or Pro Tools audio editing session, which natively handles AIFC files.
- Archiving the audio track from broadcast MPEG-2 recordings into an uncompressed AIFC format for long-term preservation in a professional audio library.
- Pulling a music or sound design track from an MPG video master to deliver to a mastering engineer who requires uncompressed AIFF/AIFC source files.
- Separating the MP2 audio from legacy VHS-to-MPEG capture files and converting it to lossless PCM so it can be edited without further generational quality loss in an Apple-based post-production pipeline.
- Extracting interview or voiceover audio from an MPEG-1 encoded video (e.g., from an old CD-ROM or kiosk archive) to integrate into a modern podcast or documentary project requiring high-fidelity source files.
Frequently Asked Questions
No. The source audio in an MPG file is already compressed as MP2 (or sometimes MP3 or AAC), which is a lossy format. Converting it to AIFC with PCM encoding captures that audio in an uncompressed form, but it cannot recover detail that was discarded during the original MP2 encoding. You get a lossless container holding a lossy-origin signal — which is still valuable for editing, since no further compression artifacts will accumulate in your workflow.
The MPG's audio track is stored as compressed MP2, which typically runs at 192–384 kbps. AIFC using pcm_s16be stores every audio sample uncompressed at 16 bits per sample, resulting in a bitrate of roughly 1,411 kbps for stereo 44.1 kHz audio — about four to seven times larger. This is expected and is the nature of converting from a compressed format to uncompressed PCM.
AIFC supports several codecs including pcm_s16be, pcm_s24be, pcm_s32be, pcm_f32be, pcm_f64be, pcm_alaw, and pcm_mulaw. The default pcm_s16be (16-bit signed, big-endian PCM) is chosen because it provides standard CD-quality audio at a widely compatible bit depth, and big-endian byte ordering is native to the AIFC/AIFF format specification. If your workflow requires higher fidelity, you can manually substitute -c:a pcm_s24be or -c:a pcm_s32be in the FFmpeg command.
Most metadata stored in an MPG container — such as program stream timing information, MPEG system headers, and any embedded video metadata — is not applicable to an audio-only AIFC file and will be discarded. Basic audio properties like sample rate and channel count are preserved, but ID3-style tags or broadcast metadata fields typically present in MPEG files are not automatically mapped to AIFC's metadata fields by FFmpeg.
Replace the -c:a pcm_s16be flag with your desired codec. For example, use -c:a pcm_s24be for 24-bit audio, which is common in professional audio work, or -c:a pcm_alaw for telephony-compatible mu-law encoding. Note that the -b:a 128k bitrate flag is relevant only for compressed codecs and has no effect when using uncompressed PCM codecs like pcm_s16be — you can omit it for PCM output without any impact on quality.
Yes. On Linux or macOS, you can run a shell loop such as: for f in *.mpg; do ffmpeg -i "$f" -c:a pcm_s16be -b:a 128k "${f%.mpg}.aifc"; done. On Windows Command Prompt, use: for %f in (*.mpg) do ffmpeg -i "%f" -c:a pcm_s16be -b:a 128k "%~nf.aifc". This applies the same extraction and PCM encoding to every MPG file in the directory, which is especially useful for archiving a collection of VCD or broadcast recordings.
Technical Notes
The AIFC format uses big-endian byte ordering for its PCM data, which is a defining characteristic inherited from the original AIFF specification designed for Motorola 68k-era Apple hardware. When FFmpeg writes pcm_s16be into an AIFC container, the byte order is handled correctly and the file will be recognized natively by macOS Core Audio, Logic Pro, Pro Tools, and other Apple-ecosystem tools without conversion. One important limitation to note: MPG files do not support multiple audio tracks, subtitles, or chapter markers, so none of those concerns are relevant to this conversion. The source MPG video stream is completely ignored and not remuxed — FFmpeg handles this automatically when writing to an audio-only container like AIFC. If your MPG source was encoded at a non-standard sample rate (uncommon but possible in some MPEG-1 files), FFmpeg will preserve that sample rate in the AIFC output. The -b:a 128k flag in the base command is technically a no-op for uncompressed PCM codecs but is harmless; the actual output bitrate is determined entirely by sample rate, bit depth, and channel count.