Convert WebM to WAV — Free Online Tool
Convert WebM video files to WAV audio by extracting and decoding the Opus or Vorbis audio stream into uncompressed PCM — ideal for audio production workflows that require lossless, broadcast-ready WAV files. This tool runs entirely in your browser with no file uploads required.
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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
WebM files typically carry audio encoded with Opus or Vorbis — both lossy codecs optimized for web streaming. During this conversion, FFmpeg discards the VP9 video stream entirely and decodes the compressed audio stream into raw 16-bit signed little-endian PCM samples, which are then wrapped in a WAV container. This means the audio is fully decompressed from its lossy Opus or Vorbis encoding into an uncompressed waveform. There is a one-time quality cost at the point of decoding the lossy source — the resulting WAV will be large and uncompressed, but it cannot recover detail that was lost when the WebM was originally encoded. No re-encoding from one lossy format to another occurs here; the output is strictly uncompressed PCM.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser version, this runs via FFmpeg.wasm compiled to WebAssembly, executing the same logic as the desktop binary without any server-side processing. |
-i input.webm
|
Specifies the input WebM file. FFmpeg reads the container, identifies the VP9 video stream and the Opus or Vorbis audio stream, and prepares to process them according to the output options that follow. |
-c:a pcm_s16le
|
Instructs FFmpeg to decode the WebM's Opus or Vorbis audio and re-encode the audio stream as 16-bit signed little-endian PCM — the standard uncompressed format used in WAV files and the default expected by most audio production software and broadcast workflows. |
output.wav
|
Defines the output filename and container. The .wav extension tells FFmpeg to write a RIFF WAV container, which automatically limits output to audio-only streams and causes the VP9 video track from the WebM to be silently discarded. |
Common Use Cases
- Extract the audio from a WebM screen recording or lecture capture to import into a DAW like Audacity, Adobe Audition, or Logic Pro, which work natively with WAV files
- Prepare a WebM podcast recording or interview — originally captured for web distribution — for broadcast submission, which often mandates uncompressed WAV masters
- Pull audio from a WebM game clip or cutscene to use as a sound effect or music stem in a video editing project that requires PCM-compatible source files
- Convert a WebM music video downloaded from a web source into WAV so it can be imported into CD mastering or vinyl production software that rejects compressed audio containers
- Decode a WebM voiceover recording from a browser-based recorder into WAV for transcription services or speech analysis tools that require uncompressed PCM input
- Archive a WebM web conference recording as a WAV audio file for long-term storage in an uncompressed format with maximum compatibility across future software
Frequently Asked Questions
Not in the original sense — Opus is a lossy codec, so some audio information was discarded when the WebM was first encoded. The WAV output is an exact, uncompressed decode of whatever Opus preserved, so no further quality is lost in this conversion step. What you get is a lossless container holding a lossy-sourced signal. If the original recording was high quality, the resulting WAV will sound indistinguishable to most listeners.
WebM's Opus codec achieves very high compression — a typical 128 kbps Opus stream uses roughly 10–15× less space than the equivalent uncompressed 16-bit 44.1 kHz stereo PCM in WAV. When FFmpeg decodes the Opus stream into raw PCM samples for the WAV container, none of that compression is retained, so file sizes commonly jump from a few megabytes to hundreds of megabytes for longer recordings. This is expected and unavoidable with uncompressed WAV.
FFmpeg decodes the Opus stream to PCM and writes it directly to WAV — it does not re-encode to a second lossy format, so there is no additional generation loss. The command uses pcm_s16le as the output codec, which is a lossless representation of the decoded signal. The only quality ceiling is whatever Opus introduced when the WebM was originally created.
The VP9 video stream is silently dropped. FFmpeg's default behavior when writing to a WAV container is to output only audio, since WAV cannot hold video. No explicit flag is needed to discard the video — the WAV format specification itself limits output to audio-only streams.
Replace pcm_s16le in the command with pcm_s24le for 24-bit depth or pcm_s32le for 32-bit integer depth: ffmpeg -i input.webm -c:a pcm_s24le output.wav. Higher bit depths increase file size but can be useful when the WAV will be processed further in a DAW, as they preserve headroom for mixing operations. For floating-point output, use pcm_f32le instead.
Yes — in a Bash shell you can loop over files: for f in *.webm; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.webm}.wav"; done. On Windows Command Prompt, use: for %f in (*.webm) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". The browser-based tool processes one file at a time, so the FFmpeg command is the recommended approach for bulk conversions or files over 1 GB.
Technical Notes
The default output codec, pcm_s16le, produces 16-bit signed little-endian PCM — the same sample format used in standard audio CDs and compatible with virtually every audio application, operating system, and hardware device. If your WebM source contains Vorbis audio rather than Opus, FFmpeg handles both transparently with the same command; the decoder is selected automatically based on the stream's codec tag. WAV files have a theoretical 4 GB file size ceiling due to the 32-bit chunk size field in the RIFF header; for very long recordings at high sample rates, FFmpeg will either warn or switch to a W64 or RF64 variant — something to watch for with files approaching that limit. Metadata such as title, artist, or comment tags stored in the WebM container are not preserved in the WAV output, as the standard WAV RIFF header has very limited metadata support. If metadata retention matters, consider embedding tags with a tool like bwfmetaedit after conversion. Subtitle tracks, chapter markers, and additional audio tracks present in the WebM are all discarded, since WAV supports none of these features.