Extract Audio from WebM to WAV — Free Online Tool
Extract audio from a WebM file and save it as a WAV — converting the compressed Opus or Vorbis audio stream into uncompressed PCM (pcm_s16le) audio. Ideal when you need a lossless, universally compatible audio file from a web-optimized video, such as a browser recording or YouTube download.
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
WebM files typically carry audio encoded with Opus or Vorbis — both lossy, compressed codecs optimized for streaming over the web. WAV, on the other hand, stores audio as raw uncompressed PCM data. This conversion does two things: it discards the video stream entirely (no re-encoding of video, it's simply dropped), and it decodes the Opus or Vorbis audio and re-encodes it into 16-bit little-endian PCM (pcm_s16le) — the standard uncompressed format used in WAV files. Because the source audio is lossy, this is a lossy-to-lossless conversion: the WAV output will be uncompressed and of maximum fidelity relative to the WebM source, but any quality lost during the original Opus/Vorbis encoding cannot be recovered. The result is a much larger file than the WebM, but one that is readable by virtually every audio tool, DAW, or platform without requiring codec support.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In this browser-based tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no files leave your device. The same command works identically in a desktop FFmpeg installation. |
-i input.webm
|
Specifies the input WebM file. FFmpeg will parse its Matroska-based container structure, detecting the video stream (typically VP9) and audio stream (typically Opus or Vorbis) within. |
-vn
|
Disables video output entirely — the VP9 (or other) video stream in the WebM is discarded and not written to the output file. Since WAV is an audio-only format, this flag ensures FFmpeg doesn't attempt to handle the video stream at all. |
-c:a pcm_s16le
|
Decodes the WebM's Opus or Vorbis audio stream and re-encodes it as signed 16-bit little-endian PCM — the standard uncompressed audio codec used in WAV files. This produces a file that is universally compatible with audio software, DAWs, and broadcast workflows. |
output.wav
|
Defines the output filename and format. The .wav extension tells FFmpeg to wrap the pcm_s16le audio data in a RIFF/WAV container, the format expected by virtually all audio applications on Windows, macOS, and Linux. |
Common Use Cases
- Extracting audio from a WebM screen recording (e.g., from OBS or a browser API capture) to import into a DAW like Audacity or Adobe Audition for editing
- Converting a downloaded WebM lecture or conference recording into WAV for transcription software that requires uncompressed PCM input
- Preparing audio from a WebM video for use in a broadcast or post-production pipeline that mandates WAV as the delivery format
- Stripping audio from a WebM animation or motion graphic to use as a standalone sound effect or music asset in a video game engine
- Converting a WebM podcast or interview recording to WAV before mastering and distributing to podcast platforms that prefer uncompressed source files
- Archiving the audio track from a web-sourced WebM clip in an uncompressed format for long-term preservation without container dependency
Frequently Asked Questions
Yes and no. The WAV will be a perfect uncompressed representation of what the Opus or Vorbis codec preserved — but those codecs are lossy, meaning some audio detail was discarded when the WebM was originally created. Converting to WAV does not restore that lost information. What you get is the highest-fidelity version of what's in the WebM, stored in an uncompressed PCM format, but it will not sound better than the source.
This is expected. WebM uses the Opus codec, which achieves high-quality audio at bitrates as low as 96–128 kbps through perceptual compression. WAV with pcm_s16le stores every audio sample as raw 16-bit data — typically around 1,411 kbps for stereo CD-quality audio (44.1 kHz). A 10-minute WebM audio track at 128 kbps might be around 10 MB, while the equivalent WAV could be 100 MB or more.
Yes. Unlike some container-to-container conversions where the audio stream is simply copied without re-encoding, this conversion must fully decode the Opus or Vorbis bitstream and then re-encode it as uncompressed PCM. WAV does not support Opus or Vorbis — it only understands PCM and a small number of legacy codecs. This decode-and-re-encode step is what causes the file size increase.
Probably not fully. WebM uses Matroska-style metadata tags, while WAV uses INFO chunks or ID3 tags depending on the application. FFmpeg does attempt to map common metadata fields, but WAV has limited and inconsistently supported metadata capabilities. If metadata preservation is critical, check the output with a tag editor — you may need to add tags manually after conversion.
Replace pcm_s16le in the FFmpeg command with your desired codec: use pcm_s24le for 24-bit signed PCM, pcm_s32le for 32-bit signed PCM, or pcm_f32le for 32-bit floating-point PCM. For example: ffmpeg -i input.webm -vn -c:a pcm_s24le output.wav. Higher bit depths won't recover detail lost in the original Opus encoding, but pcm_f32le is useful if the WAV will be processed further in a floating-point audio pipeline.
Yes, using a shell loop. On Linux or macOS: for f in *.webm; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.webm}.wav"; done. On Windows Command Prompt: for %f in (*.webm) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". This applies the same conversion — drop video, decode Opus/Vorbis, output uncompressed WAV — to every WebM file in the current directory.
Technical Notes
WebM audio is almost always encoded with Opus (the default in modern WebM) or the older Vorbis codec. Both are variable-bitrate lossy codecs, so the source audio quality depends entirely on the bitrate used when the WebM was created — typically between 64 kbps and 320 kbps. The output WAV uses pcm_s16le: signed 16-bit samples, little-endian byte order, which is the standard for CD-quality audio and is natively readable on Windows, macOS, Linux, and virtually every DAW and audio editor. The sample rate and channel count from the WebM source are preserved unless you explicitly specify -ar (audio sample rate) or -ac (audio channels) flags. If the WebM contains multiple audio tracks (a feature WebM supports), FFmpeg will extract the first audio track by default; use -map 0:a:1 to target a specific track. WAV does not support subtitles, chapters, or multiple audio tracks — all of which WebM can carry — so that metadata is simply dropped. The maximum practical WAV file size is just under 4 GB due to the 32-bit chunk size field in the RIFF header; for very long recordings, consider using the W64 or RF64 format instead.