Convert WebM to WAV Online
Extract and decode audio from WebM files to uncompressed WAV. Ideal for screen recordings, browser audio, and podcast post-production. Free, in your browser.
.webm · up to 100 MB
What you can do
WebM to WAV: screen and browser recording audio ready to edit
Opus and Vorbis
Correctly decodes both Opus and Vorbis audio inside the WebM container.
100% private
Screen recordings contain sensitive material. Your WebM never leaves your device.
DAW and transcription
Uncompressed WAV is the ideal format for professional editing and speech recognition.
Browser recordings
Compatible with WebM files generated by the Chrome and Firefox MediaRecorder API.
How it works
Three steps, no hassle
Upload your WebM file
Drag or select your .webm file. Up to 500 MB. No signup or installs.
WebM audio decoding
Opus or Vorbis audio from the WebM container is decoded to uncompressed PCM in your browser.
Download your WAV
Uncompressed PCM audio ready for editing in any DAW or professional audio editor.
FAQ
Got questions?
Technically speaking, the resulting WAV introduces no new quality loss beyond what already exists in the original WebM audio. WebM uses lossy audio codecs (Opus or Vorbis), so one generation of loss already exists from when the WebM file was created. Converting to WAV simply decodes those codecs to their underlying PCM representation, without adding new artifacts. The resulting WAV is mathematically equivalent to the PCM that any reference Opus or Vorbis decoder would produce: it faithfully captures the signal as encoded in WebM, including the original compression artifacts, but introduces none of its own. If the WebM audio was recorded at high quality (Opus at 128 kbps or above, Vorbis q5 or above), the resulting WAV will be of very high perceptual quality for professional editing.
It depends on the specific WebM file. The WebM container (developed by Google from 2010 as part of the VP8/WebM project) officially supports two audio codecs: Vorbis (the original codec in the first WebM specification, 2010) and Opus (added to the WebM specification in 2013 when Google adopted Opus as the preferred codec). Recordings made with the MediaRecorder API in modern browsers (Chrome, Firefox, Edge) mostly use Opus, as it is the default codec for MediaRecorder in most implementations. WebM files from downloaded YouTube content use Vorbis (for standard qualities) or Opus (for high qualities). The tool automatically detects the audio codec in the container and applies the correct decoder without manual configuration.
Opus is technically superior to Vorbis at virtually all bitrates and content types. Opus was developed by Xiph.org and Mozilla, with RFC 6716 published in September 2012 by the IETF. Unlike Vorbis, which is a pure transform coding codec (MDCT), Opus combines SILK (a CELP codec originally designed for Skype, later licensed to Xiph) for low-bitrate voice audio, and CELT (Constrained Energy Lapped Transform, also from Xiph) for music and full-band audio. This hybrid architecture makes Opus exceptionally efficient: at 64 kbps Opus outperforms Vorbis at 160 kbps in voice quality tests, and at 128 kbps Opus is practically indistinguishable from the original in most musical content. If your WebM uses Opus, the resulting WAV will be high quality even if the original bitrate was modest.
Modern web browsers implement the MediaRecorder API (part of the W3C Media Capture and Streams specification, standardized around 2013–2015) to record audio and video directly from JavaScript. This API was designed to produce WebM as output because WebM is the only open-source video format with broad native browser support that also includes audio: VP8 or VP9 for video and Opus for audio. Browser screen recording tools like Screenity, Loom (browser mode), OBS (browser source), and web video calling apps that offer local recording (Jitsi, BigBlueButton) typically produce WebM. If you need audio from a browser screen recording for post-production (voice editing, noise removal, transcription), converting the WebM to WAV is the standard first step in the workflow.
Significantly larger. A WebM with Opus audio at 128 kbps occupies approximately 0.96 MB per minute. The equivalent uncompressed WAV (44.1 kHz, 16-bit, stereo) occupies about 10 MB per minute. The typical expansion factor is 8 to 12 times. For a 30-minute screen recording in WebM (≈29 MB of audio), the resulting WAV will take about 300 MB. If space is a concern, consider converting to FLAC (lossless PCM compression, 50–60% reduction) or keeping OGG/Opus for archiving and converting to WAV only when you need to edit in a DAW.
Professional use cases are varied. First, podcast post-production: interviews recorded with web video call tools (Zencastr, Riverside.fm, SquadCast) that produce WebM as local backup, converted to WAV for editing in Adobe Audition, iZotope RX, or Hindenburg. Second, audio transcription: automatic transcription services like OpenAI Whisper, Otter.ai, or Descript accept WAV as the preferred format for maximum speech recognition model accuracy. Third, noise reduction processing: tools like iZotope RX 10 Advanced or CEDAR DNS operate on uncompressed PCM. Fourth, video sync in a DAW: importing the audio from a screen recording as WAV into DaVinci Resolve, Premiere Pro, or Final Cut Pro to replace the original video audio or for detailed editing. Fifth, broadcast delivery requirements: many TV networks and professional podcast platforms specify WAV (or AIFF) as the standard delivery format.
Convert WebM to WAV: decode Opus and Vorbis audio for professional editing
WebM is an open-source multimedia container format developed by Google from 2010, designed specifically for the web as a patent-free alternative to H.264/MP4. The WebM project emerged when Google acquired On2 Technologies in 2010 and released the VP8 codec as free software, combining it with Vorbis audio (already existing at Xiph.org) in the Matroska-derived container named WebM. In 2013, the WebM specification was updated to support VP9 (VP8's successor) and the Opus codec as the preferred audio for the container. Opus was jointly developed by Xiph.org and Mozilla, with RFC 6716 published in September 2012: it is a hybrid codec combining SILK (designed for Skype) for low-latency voice, and CELT (from Xiph.org) for full-band audio and music. The choice of Opus as the default audio codec in WebM and in the browser MediaRecorder API has solid technical grounding: Opus was adopted by the IETF as the mandatory codec for WebRTC (RFC 7742, 2016), making it the universal codec for real-time audio communications in browsers. WAV, in turn, is the container for uncompressed PCM audio developed by Microsoft and IBM in 1991, and remains the universal interchange format in professional audio production for its simplicity (no decoding required) and precision (sample-level random access).
Extracting audio from WebM to WAV is a two-stage process: first, demuxing the WebM container (which is technically a simplified subset of the Matroska container, restricted to VP8/VP9 for video and Opus/Vorbis for audio) to isolate the compressed audio stream, then decoding the audio codec (Opus or Vorbis) to PCM. The Opus decoder (libopus, the Xiph.org reference implementation) produces PCM at 48 kHz as its native sample rate: Opus always works internally at 48 kHz regardless of the original audio's sample rate, an important difference from Vorbis which preserves the original frequency. The Vorbis decoder produces PCM at the sample rate specified in the stream header (typically 44.1 kHz for music content, or 48 kHz for web content). Chrome's MediaRecorder API produces WebM with Opus audio at 48 kHz and variable bitrate by default; Firefox implements MediaRecorder with support for both Opus and Vorbis depending on configuration. The resulting WAV will have the sample rate of the decoded PCM: 48 kHz if the audio was Opus, or 44.1 or 48 kHz if it was Vorbis according to the original configuration.
Audio post-production workflows requiring WebM-to-WAV conversion are particularly prevalent in the digital content creation ecosystem. Remote podcast recording platforms like Zencastr, Riverside.fm, SquadCast, and Cleanfeed record each participant's audio locally in the browser using the MediaRecorder API, producing WebM files that are then uploaded to a server for editing. These WebM files are the highest-quality audio source available (recorded directly from the microphone, without VoIP compression), and converting them to WAV is the first step before editing in Adobe Audition, iZotope RX, or any professional DAW. Another frequent case is automatic audio transcription: speech recognition models like OpenAI Whisper, which analyzes audio segments in WAV/MP3, offer better accuracy with 16 kHz mono WAV than with compressed audio. Google Meet, Microsoft Teams, or Zoom meeting recordings exported as WebM can be converted to WAV for detailed transcription or for archiving in document management systems requiring uncompressed audio formats. Convertir.ai performs all WebM→WAV processing in the browser via WebAssembly, ensuring that recording contents — which frequently include confidential conversations or unpublished material — are never transmitted to any external server.