Convert MPEG to WAV — Free Online Tool
Extract and convert the audio track from an MPEG video file into a WAV file using uncompressed 16-bit PCM audio. This tool strips the MP2 or MP3 audio from the MPEG container and decodes it into a lossless PCM WAV stream, making it ideal for audio editing workflows that require uncompressed source material.
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FFmpeg Command
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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
MPEG files typically carry audio encoded in MP2 (MPEG-1 Audio Layer II) or occasionally MP3, embedded alongside MPEG-1 or MPEG-2 video. During this conversion, FFmpeg discards the video stream entirely and decodes the compressed audio into raw 16-bit signed little-endian PCM samples, which are then wrapped in a WAV container. This is a full decode-and-re-encode operation on the audio — the lossy MP2/MP3 compression is unwrapped and the decompressed audio data is written out as uncompressed PCM. The resulting WAV file is larger than the original audio track but introduces no additional generation loss beyond what was already present in the MPEG's lossy encoding. The video track is dropped because WAV is a pure audio container with no support for video streams.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles the demuxing of the MPEG container, decoding of the compressed audio stream, and writing of the output WAV file. |
-i input.mpeg
|
Specifies the input MPEG file. FFmpeg reads the MPEG-1 or MPEG-2 container, identifies the video stream (mpeg1video or mpeg2video) and the audio stream (typically MP2), and makes both available for processing. |
-c:a pcm_s16le
|
Instructs FFmpeg to decode the MPEG's compressed MP2 or MP3 audio and re-encode the output as 16-bit signed little-endian PCM — the standard uncompressed audio format used in WAV files and CD audio. This produces the highest-fidelity, universally compatible WAV output. |
output.wav
|
Defines the output file name and format. The .wav extension causes FFmpeg to wrap the PCM audio data in a RIFF WAV container. No video stream is included because WAV supports audio only, so FFmpeg automatically drops the video from the MPEG source. |
Common Use Cases
- Extracting dialogue or narration from legacy broadcast MPEG recordings to import into a DAW like Audacity or Adobe Audition for editing
- Pulling the audio from a DVD-compatible MPEG file to use as a clean source for audio restoration or noise reduction tools that require uncompressed input
- Converting MPEG news broadcast recordings into WAV for transcription software that performs better with uncompressed PCM audio
- Archiving the audio track from old MPEG home video captures into an uncompressed format before further processing or compression to a modern codec like AAC or FLAC
- Extracting the MP2 audio from an MPEG stream and decoding it to WAV so it can be ingested by professional broadcast tools that do not accept MP2 input directly
- Preparing audio from MPEG training or presentation videos for use in e-learning platforms that require WAV uploads
Frequently Asked Questions
The WAV output will be a faithful, bit-accurate decode of the audio that was stored in the MPEG file. However, the original MPEG audio (typically MP2 or MP3) was already lossy-compressed when the MPEG was created, so any artifacts introduced at that stage are baked in. Converting to WAV does not recover lost audio information — it simply decodes what is there into an uncompressed form with no further quality degradation.
MPEG files use lossy compression for both video and audio, storing audio as compact MP2 or MP3 data. WAV with PCM encoding stores every audio sample as raw, uncompressed data — typically 16 bits per sample at the original sample rate. A one-minute MPEG audio track at 192 kbps might be around 1.4 MB, while the equivalent 16-bit stereo PCM WAV at 44.1 kHz would be roughly 10 MB. This size increase is expected and is the nature of uncompressed audio.
Yes. By default, FFmpeg preserves the sample rate and channel count from the source MPEG audio during decoding. If your MPEG audio is 48 kHz stereo (common in broadcast MPEG-2), the output WAV will also be 48 kHz stereo. If you need to resample or change the channel layout, you would need to add extra FFmpeg flags such as -ar and -ac to the command.
The command uses -c:a pcm_s16le, which produces standard 16-bit signed PCM. If you need 24-bit audio (common in professional audio workflows), change the codec flag to -c:a pcm_s24le. For 32-bit floating-point PCM, use -c:a pcm_f32le. The rest of the command stays the same: ffmpeg -i input.mpeg -c:a pcm_s24le output.wav.
Yes. On Linux or macOS you can run a shell loop: for f in *.mpeg; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.mpeg}.wav"; done. On Windows Command Prompt, use: for %f in (*.mpeg) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This processes every MPEG file in the current directory and outputs a corresponding WAV file.
Yes, the video stream is dropped entirely because WAV is a pure audio container — it has no mechanism to store video data. Only the audio track from the MPEG is decoded and written to the WAV file. If you need to keep the video, you should convert to a format like MP4 or MKV instead. If you want to extract both audio and video separately, you would run two separate FFmpeg commands.
Technical Notes
MPEG files (using the .mpeg, .mpg, or .vob extension) most commonly carry audio encoded as MP2 (MPEG-1 Audio Layer II) at bitrates between 128 kbps and 384 kbps, though some MPEG-1 files use MP3 and some broadcast MPEG-2 streams include AAC. FFmpeg auto-detects the audio codec from the stream headers, so the same command works regardless of which specific audio codec the MPEG uses. The output codec pcm_s16le is 16-bit signed little-endian PCM — the standard encoding used in CD audio and the most universally compatible WAV variant, accepted by virtually all audio editors, DAWs, and media players. WAV files produced this way contain a standard RIFF header followed by raw PCM data; metadata from the MPEG source (such as title or date tags) is generally not preserved, as MPEG containers carry minimal metadata and WAV's metadata support (via INFO chunks or ID3 tags) is limited and inconsistently implemented across applications. There is no video quality parameter relevant to this conversion since the video is discarded, and WAV's PCM encoding has no quality setting — the bit depth is the only meaningful quality control, defaulting here to 16-bit.