Extract Audio from MPG to WAV — Free Online Tool
Extract the audio track from an MPG video file and save it as an uncompressed WAV file. This tool decodes the MP2 or AAC audio stream embedded in the MPEG-1/2 container and re-encodes it to 16-bit PCM WAV — the standard lossless format for audio editing, archiving, and broadcast 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
MPG files store audio using lossy codecs — most commonly MPEG Layer 2 (MP2) in broadcast and DVD/VCD sources, though AAC and MP3 also appear. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio into raw PCM samples, which are then written into a WAV container using the pcm_s16le codec (signed 16-bit little-endian PCM). This is a full decode-and-re-encode of the audio — not a stream copy — because WAV's default PCM format is fundamentally different from the lossy codecs used in MPG. The result is an uncompressed WAV file that is lossless in the sense that no further generation loss is introduced beyond what was already present in the original MPG audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles all the demuxing, decoding, and re-encoding work in this conversion. When running in the browser, this is executed via FFmpeg.wasm, a WebAssembly port that processes your file entirely locally. |
-i input.mpg
|
Specifies the input file — an MPG container holding MPEG-1 or MPEG-2 video alongside a compressed audio stream (typically MP2 in VCD/broadcast sources). FFmpeg demuxes both streams from the MPEG program or transport stream structure. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the mpeg1video or mpeg2video stream in the MPG file. Without this flag, FFmpeg would attempt to include video in the output, which WAV cannot contain and would cause an error. |
-c:a pcm_s16le
|
Decodes the MPG's compressed audio (typically MP2) and re-encodes it as signed 16-bit little-endian PCM — the standard uncompressed audio codec for WAV files. This produces maximum compatibility across DAWs, editors, and operating systems. |
output.wav
|
Specifies the output file. The .wav extension tells FFmpeg to use the WAV container format, which wraps the uncompressed pcm_s16le audio data with a RIFF/WAV header that encodes sample rate, bit depth, and channel count. |
Common Use Cases
- Extracting the audio from a VCD or DVD MPEG source to import into a DAW like Audacity or Adobe Premiere for audio cleanup or re-mixing
- Pulling the MP2 audio track from a broadcast-captured MPG file and converting it to WAV for delivery to a radio station or post-production house that requires uncompressed audio
- Archiving the audio content of legacy MPG video recordings (home videos, news captures) in a non-proprietary, widely-supported format before the original files degrade
- Extracting a narration or dialogue track from an MPEG-1 training video to re-use or re-dub in a new production
- Converting MPG audio to WAV as a lossless intermediate before further encoding to MP3, AAC, or OGG to avoid double-lossy compression artifacts
- Preparing audio from an MPG source for waveform analysis, transcription, or speech recognition tools that require uncompressed PCM input
Frequently Asked Questions
No — the WAV output cannot recover quality that was lost when the audio was originally encoded into the MPG file as MP2 or AAC. What it does do is stop any further quality degradation: once the audio is in uncompressed PCM WAV format, you can edit, process, or re-encode it without introducing additional lossy-compression artifacts. Think of it as capturing a lossless snapshot of whatever quality exists in the source.
MPG stores audio using lossy compression — MP2 typically runs at 192–384 kbps, achieving significant size reduction. WAV with pcm_s16le is completely uncompressed: a stereo 44.1kHz file uses about 10 MB per minute. A 700MB VCD-sourced MPG might yield a WAV audio track that is several times larger than the original audio-only portion, because uncompressed PCM has no compression at all. This is expected and normal.
Yes. FFmpeg reads the sample rate and channel count directly from the decoded MPG audio stream and writes them into the WAV header without modification. So if your MPG contains stereo MP2 audio at 48kHz (common in broadcast MPEG-2), the output WAV will also be stereo 48kHz PCM. No resampling or channel remapping occurs unless you explicitly add flags to the command.
Replace pcm_s16le with pcm_s24le in the FFmpeg command: ffmpeg -i input.mpg -vn -c:a pcm_s24le output.wav. This writes 24-bit little-endian PCM, which is preferred for professional audio mastering. Other options include pcm_s32le for 32-bit integer or pcm_f32le for 32-bit float. Note that since the MPG source audio is lossy, the extra bit depth captures the full decoded dynamic range but does not recover information that lossy compression discarded.
Yes. On Linux or macOS, you can loop over files in a directory: for f in *.mpg; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.mpg}.wav"; done. On Windows Command Prompt, use: for %f in (*.mpg) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". Each MPG file will produce a corresponding WAV file with the same base name, processed sequentially.
FFmpeg will exit with an error stating it cannot find a suitable audio stream. Some MPEG-1/2 files — particularly those encoded for silent video or with stripped audio — genuinely contain no audio. You can verify whether audio exists by running ffprobe input.mpg and checking the stream list before attempting the extraction.
Technical Notes
MPG is a container defined by the MPEG-1 and MPEG-2 standards, with audio most commonly encoded as MPEG Layer 2 (MP2) — a precursor to MP3 that was mandated for VCD and is standard in DVB broadcast. The FFmpeg flag -vn explicitly drops all video streams, preventing FFmpeg from attempting to transcode the mpeg1video or mpeg2video stream into the WAV container (which cannot hold video). The output codec pcm_s16le is the canonical WAV audio format: signed 16-bit samples in little-endian byte order, compatible with virtually every audio application on every platform. WAV metadata support is limited — ID3-style tags from the MPG are not automatically carried over, and WAV's own INFO chunk metadata support is minimal, so title, artist, or chapter information will not be preserved. If the MPG source was interlaced broadcast content captured at 48kHz, the WAV will reflect that sample rate, which is fine for professional use but worth noting if your downstream tools assume 44.1kHz. Files over 4GB would exceed the standard WAV file size limit (due to its 32-bit chunk size field), but for typical audio-only extractions from MPG sources this is unlikely to be a concern.