Extract Audio from VOB to WAV — Free Online Tool
Extract uncompressed PCM audio from VOB files — the raw video container used on DVD discs — and save it as a WAV file. This tool strips the AC3 (Dolby Digital) or other audio streams from the MPEG-2 multiplexed container and decodes them to lossless 16-bit PCM, giving you a clean, universally compatible audio file ideal for editing or archiving.
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
VOB files store multiplexed streams of MPEG-2 video, AC3 (Dolby Digital) audio, and sometimes subtitles in a single container. This conversion discards the video stream entirely and decodes the audio — typically AC3 at 192–448 kbps — into uncompressed PCM (pulse-code modulation) audio at 16-bit depth, then wraps it in a WAV container. Because AC3 is a lossy format, the WAV output is not bit-for-bit identical to the original recording, but it is a lossless representation of the decoded AC3 audio with no further quality degradation introduced. Multi-channel audio (e.g., 5.1 surround) from the VOB is preserved in channel count unless you explicitly downmix it.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles demuxing the VOB container, decoding the AC3 audio stream, and encoding the output WAV file — all running locally in your browser via WebAssembly. |
-i input.vob
|
Specifies the input VOB file. FFmpeg reads the MPEG program stream container, identifying the multiplexed MPEG-2 video, AC3 audio, and any subtitle streams within the DVD-format sectors. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the MPEG-2 video stream. This is essential because WAV is an audio-only container and cannot hold video data. |
-c:a pcm_s16le
|
Decodes the AC3 (Dolby Digital) audio from the VOB and re-encodes it as 16-bit signed little-endian PCM — the standard uncompressed audio format used in WAV files and compatible with virtually every audio application and device. |
output.wav
|
Sets the output filename and tells FFmpeg to wrap the PCM audio in a WAV (Waveform Audio File Format) container, which includes the necessary RIFF headers that allow media players, DAWs, and audio editors to read the file correctly. |
Common Use Cases
- Rip the audio commentary track from a DVD VOB file to preserve it as a high-quality archive for editing or transcription
- Extract a film's dialogue or soundtrack from a backup VOB rip to use in a video editing project that requires an uncompressed audio source
- Convert DVD concert or live performance audio from AC3-encoded VOB files into WAV for import into a DAW like Pro Tools or Audacity
- Prepare extracted DVD audio for audio restoration work, where the uncompressed WAV format is required for noise reduction or remastering plugins
- Extract spoken-word content from educational or training DVDs into WAV format for use in transcription software
- Archive the audio from a home-burned DVD of a family event as a standalone WAV file before the disc degrades
Frequently Asked Questions
The WAV file is a lossless decode of the AC3 (Dolby Digital) audio that was already encoded on the DVD — no additional quality loss occurs during this conversion. However, because AC3 is itself a lossy codec, the WAV does not represent perfectly pristine original audio; it reflects whatever compression artifacts existed in the AC3 stream. Think of it as a perfect snapshot of the lossy source, not a restoration of it.
Yes, by default FFmpeg decodes and preserves all audio channels from the AC3 stream, so a 5.1 surround track will produce a 6-channel WAV file. Most standard media players and DAWs support multi-channel WAV files, though some consumer software may only play back the front stereo pair. If you need a stereo downmix, you can add '-ac 2' to the command before the output filename.
VOB files from DVDs often contain multiple audio streams for different languages or commentary. You can target a specific track by adding '-map 0:a:1' (for the second audio track, using zero-based indexing) to the FFmpeg command before the output filename. Run 'ffmpeg -i input.vob' first to list all available streams and identify which index corresponds to your desired language track.
The VOB's audio is stored as compressed AC3, which typically uses 192–448 kbps to encode audio. The WAV output uses uncompressed 16-bit PCM, which for stereo audio at 48 kHz (the standard DVD sample rate) uses approximately 1,536 kbps — three to eight times more data per second. A two-hour DVD's audio track, for example, can easily produce a WAV file of 1–2 GB. This is expected and is the nature of uncompressed audio.
Yes — the default '-c:a pcm_s16le' produces 16-bit signed little-endian PCM, which is CD-quality. You can substitute 'pcm_s24le' for 24-bit output or 'pcm_s32le' for 32-bit, which may be preferable for professional audio editing workflows. However, since DVD AC3 audio is mastered at 16-bit or 20-bit depth, going beyond 24-bit offers no practical quality benefit and only increases file size.
Yes, on Linux or macOS you can run a shell loop such as 'for f in *.vob; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.vob}.wav"; done' to process every VOB file in a directory. On Windows Command Prompt, use 'for %f in (*.vob) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav"'. Note that some DVD rips split a single title across multiple numbered VOB files (e.g., VTS_01_1.vob, VTS_01_2.vob), which you may want to concatenate first using FFmpeg's concat demuxer.
Technical Notes
DVD VOB files use the MPEG program stream container format with a mandatory 2048-byte sector structure, which FFmpeg handles via the '-f vob' demuxer flag when auto-detection is insufficient. The audio codec in VOB files is almost always AC3 (Dolby Digital) at 48 kHz sample rate, though some DVDs — particularly PAL releases and some Asian titles — may include DTS, LPCM, or MPEG-1 Layer II audio streams instead. The '-vn' flag is critical here to discard the MPEG-2 video stream; without it, FFmpeg would attempt to encode or copy video into the WAV container, which is not supported and would cause an error. The output sample rate defaults to whatever the source AC3 stream uses (typically 48 kHz for DVD), and this is preserved in the WAV file without resampling. DVD chapter markers and subtitle streams are not carried over to WAV, as the format does not support them. If you are processing a concatenated DVD title spread across multiple VOB files (common for feature-length content), consider using FFmpeg's concat protocol or demuxer to stitch them before extraction to avoid silence gaps or audio discontinuities between files.