Extract Audio from MPEG to WAV — Free Online Tool
Extract audio from MPEG video files and save it as an uncompressed WAV file. This tool strips the MPEG-1/2 video stream and decodes the MP2 or MP3 audio track into raw 16-bit PCM — giving you a lossless, universally compatible WAV file ready for editing, archiving, or broadcast use.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
MPEG files typically carry MP2 (MPEG-1 Audio Layer II) or MP3 audio, which are lossy compressed formats. During this conversion, FFmpeg discards the MPEG-1 or MPEG-2 video stream entirely and decodes the compressed audio track into uncompressed PCM signed 16-bit little-endian audio — the standard encoding used in WAV files. This is a decode operation, not a remux: the compressed audio is fully decompressed into raw waveform data. The result is a WAV file with no further generation loss, though the original lossy compression artifacts from the MPEG source are baked in. File sizes will be significantly larger than the audio portion of the original MPEG, since WAV stores every sample without compression.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser-based version of this tool, FFmpeg runs via WebAssembly (ffmpeg.wasm) entirely within your browser — no files leave your machine. |
-i input.mpeg
|
Specifies the input MPEG file. FFmpeg reads the MPEG-1 or MPEG-2 container and identifies both the video stream (mpeg1video or mpeg2video) and the audio stream (typically MP2) inside it. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the MPEG-1/2 video stream. Without this flag, FFmpeg would try to include video in the output, which WAV cannot contain. |
-c:a pcm_s16le
|
Decodes the compressed MP2 (or MP3/AAC) audio from the MPEG file and re-encodes it as PCM signed 16-bit little-endian — the standard uncompressed audio codec for WAV files, compatible with virtually all audio software and devices. |
output.wav
|
Sets the output filename and container format. The .wav extension tells FFmpeg to write a RIFF WAV container, which wraps the uncompressed PCM audio data produced from the decoded MPEG audio track. |
Common Use Cases
- Extract the audio commentary or narration from an old MPEG broadcast recording for use in a video editing timeline that requires WAV input
- Recover a clean, uncompressed audio file from an MPEG training video or lecture recording for transcription software that works best with PCM WAV
- Pull the MP2 audio from a DVD-compatible MPEG file and convert it to WAV for import into a DAW like Pro Tools or Audacity, which natively handles PCM audio
- Archive the audio track from legacy MPEG home videos or digitized VHS captures as uncompressed WAV before the source media degrades further
- Extract audio from an MPEG file recorded by a broadcast encoder or set-top box for use in a radio production workflow that requires broadcast-ready WAV files
- Prepare audio from an MPEG source for forensic or scientific analysis, where uncompressed PCM ensures no additional processing artifacts are introduced
Frequently Asked Questions
No — WAV is uncompressed, but it cannot undo the lossy compression already applied to the MP2 or MP3 audio inside the MPEG file. The WAV output is an exact decoded representation of the compressed audio, artifacts and all. What you gain is a format that introduces no additional compression, making it ideal for editing or further processing without accumulating more quality loss.
MPEG files store audio in MP2 or MP3 format, which can compress audio to a fraction of its raw size. WAV with PCM encoding stores every audio sample uncompressed — typically at 1411 kbps for stereo 44.1 kHz 16-bit audio, compared to 128–192 kbps for MP2. A 100 MB MPEG file might yield a WAV file several times larger, which is expected and normal for uncompressed audio.
FFmpeg handles this automatically. Whether the MPEG file carries MP2, MP3, or AAC audio, FFmpeg decodes whichever audio codec is present and re-encodes it as 16-bit PCM into the WAV container. The command shown works regardless of which audio codec was used in the source MPEG.
You can add '-ar 44100' to set the sample rate (e.g., 44100 Hz or 48000 Hz) and '-sample_fmt s24le' along with '-c:a pcm_s24le' to switch to 24-bit depth. For example: 'ffmpeg -i input.mpeg -vn -c:a pcm_s24le -ar 48000 output.wav' produces a 24-bit, 48 kHz WAV, which is standard for professional broadcast and DAW workflows.
Yes. On Linux or macOS, you can run: 'for f in *.mpeg; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.mpeg}.wav"; done' in a terminal. On Windows Command Prompt, use: 'for %f in (*.mpeg) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav"'. This applies the same extraction to every MPEG file in the folder, which is useful for archiving large collections.
MPEG files rarely carry rich metadata, and WAV has very limited metadata support compared to formats like FLAC or MP3. FFmpeg will attempt to copy any ID3 or basic tags present in the source, but WAV's INFO chunk support is inconsistent across players and editors. If metadata preservation is critical, consider outputting to FLAC instead of WAV.
Technical Notes
MPEG container files (.mpeg, .mpg) were standardized for broadcast and disc distribution and typically carry MP2 audio — the audio format used on DVDs and digital television broadcasts — though MP3 and occasionally AAC also appear. When extracting to WAV, FFmpeg fully decodes the MP2 bitstream into PCM signed 16-bit little-endian (pcm_s16le), which is the WAV standard compatible with virtually every audio application, operating system, and hardware device. The '-vn' flag is essential here; without it, FFmpeg would attempt to encode the MPEG video into the WAV container, which WAV does not support and would cause an error. WAV files have a theoretical size limit of 4 GB due to the 32-bit size field in the RIFF header; for very long MPEG recordings, this could be a constraint at high sample rates or bit depths, though at standard 16-bit/44.1 kHz stereo PCM you can store roughly 6.5 hours before hitting the limit. No subtitle, chapter, or secondary audio track data is carried through this conversion, as both MPEG (in this context) and WAV support only a single audio stream.