Extract Audio from MPG to AAC — Free Online Tool

Extract the audio track from an MPG video file and save it as an AAC file — converting the original MP2 audio stream (common in MPEG-1/2 broadcast and DVD content) into the modern AAC format, which delivers better sound quality at equivalent or lower bit rates and is natively supported by Apple, Android, and web platforms.

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How It Works

MPG files typically carry audio encoded as MP2 (MPEG-1 Audio Layer II), a legacy lossy codec common in VCD, DVD, and broadcast video. This tool discards the video stream entirely and re-encodes only the MP2 audio track into AAC using FFmpeg's built-in AAC encoder. Because MP2 and AAC use different compression algorithms, a full decode-and-reencode is required — the MP2 audio is first decoded to raw PCM, then re-encoded to AAC at the target bit rate (default 128k). This is a lossy-to-lossy transcode, so some generational quality loss is inherent. The output is a standalone .aac file containing no video data.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg application, the open-source multimedia processing engine that handles the decoding of the MPG input and encoding of the AAC output in this conversion.
-i input.mpg Specifies the input MPG file. FFmpeg reads the MPEG-1/2 container, identifying the video stream (typically MPEG-2 video) and the audio stream (typically MP2) for processing.
-vn Disables video output entirely. This tells FFmpeg to ignore the MPEG-2 video stream from the MPG file and produce an audio-only output, which is the core behavior of an audio extraction tool.
-c:a aac Specifies the AAC encoder for the audio stream. Because the source audio is MP2 and the output must be AAC, a full re-encode is required — this flag selects FFmpeg's built-in AAC encoder to perform that conversion.
-b:a 128k Sets the target audio bit rate for the AAC output to 128 kilobits per second. At this rate, AAC delivers good perceived quality for most music and speech, and produces a file roughly 1 MB per minute of audio — significantly more efficient than the MP2 audio typically found in MPG files.
output.aac Defines the output filename and format. The .aac extension tells FFmpeg to write a raw AAC bitstream with ADTS framing, which is directly playable by most media players, mobile devices, and web browsers without requiring an additional container wrapper.

Common Use Cases

  • Ripping the audio from a VCD or DVD-sourced MPG file to create a portable AAC audio file for playback on an iPhone or iPad without carrying the full video
  • Extracting music or a soundtrack from an MPEG-2 broadcast recording to upload to an iTunes or Apple Music library, which natively supports AAC
  • Converting legacy MPG interview or lecture recordings into AAC audio for use in a podcast editing workflow where video is unnecessary
  • Stripping the audio from an MPEG-1 VCD rip to create a smaller file for archiving dialogue or commentary without storing gigabytes of video
  • Preparing audio from MPG broadcast captures for use in a mobile app or web player that requires AAC rather than MP2 for broad device compatibility
  • Extracting a musical performance recorded in MPEG-2 format to share as a standalone AAC file on streaming or social platforms that do not accept video MPG uploads

Frequently Asked Questions

Yes, some quality loss is unavoidable because this is a lossy-to-lossy transcode. The MP2 audio in the MPG file is decoded to uncompressed PCM and then re-encoded into AAC, introducing a second generation of compression artifacts. In practice, at 128k AAC the result is generally considered transparent for most listeners, but if the source MPG was encoded at a low MP2 bit rate to begin with, artifacts may be more noticeable. Using a higher AAC bit rate like 192k or 256k can help minimize perceptible degradation.
MP2 (MPEG-1 Audio Layer II) and AAC are entirely different codec formats — their compressed bitstream structures are incompatible. An AAC file can only contain audio encoded with the AAC algorithm, so the MP2 stream cannot be stream-copied into a .aac container. A full decode-and-reencode step is required, which is what the -c:a aac flag in the FFmpeg command performs.
AAC is a more efficient codec than MP2 — it achieves comparable or better perceived audio quality at lower bit rates. MP2 typically requires around 192k to 256k for good stereo quality, whereas AAC at 128k is widely considered to match or exceed that. However, since converting from MPG to AAC involves re-encoding a lossy source, the output quality is bounded by the quality of the original MP2 track, not just the target bit rate.
Replace the value after -b:a with your desired bit rate. For example, to encode at 256k instead of the default 128k, use: ffmpeg -i input.mpg -vn -c:a aac -b:a 256k output.aac. AAC supports bit rates from as low as 64k (acceptable for speech) up to 320k (high-fidelity music). For most music extracted from MPG files, 192k strikes a good balance between file size and quality.
Yes. On Linux or macOS, you can use a shell loop: for f in *.mpg; do ffmpeg -i "$f" -vn -c:a aac -b:a 128k "${f%.mpg}.aac"; done. On Windows Command Prompt, use: for %f in (*.mpg) do ffmpeg -i "%f" -vn -c:a aac -b:a 128k "%~nf.aac". This will process each MPG file in the folder and produce a corresponding .aac file, which is especially useful when ripping audio from a collection of VCD or DVD recordings.
MPG files based on the MPEG-1/2 standard have very limited metadata support compared to modern containers, so there is typically little or no embedded metadata to transfer. FFmpeg will attempt to copy any metadata tags it finds, but in practice most MPG broadcast or VCD files contain no title, artist, or album fields. If you need rich metadata in the output AAC file, you will need to add it manually using a tag editor after conversion.

Technical Notes

MPG files using the MPEG-2 standard most commonly carry MP2 audio (MPEG-1 Audio Layer II), though some MPG sources — particularly DVD rips or digitally encoded content — may contain AAC or even MP3 audio. This tool targets the typical case and re-encodes using FFmpeg's native AAC encoder, which is broadly compatible but slightly lower in quality than the optional libfdk_aac encoder available in manually compiled FFmpeg builds. The output .aac file is a raw AAC bitstream (ADTS framing), not wrapped in an MP4/M4A container — this means it is widely playable but lacks a seekable index, which can make scrubbing in some players imprecise. If you need a seekable file with better metadata support, consider outputting to .m4a instead by changing the output filename, as AAC audio in an M4A container is more compatible with iTunes, QuickTime, and most mobile players. The -vn flag ensures no video stream is written, keeping the output a pure audio file. Since MPG does not support multiple audio tracks, there is no need to specify a stream selector — the single audio track is extracted automatically.

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