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Supports: WEBM
.webm clips — screen recordings, YouTube/Twitter downloads, MediaRecorder captures, or browser voice notes. Batch conversion is supported and runs on our servers.pcm_s16le (16-bit signed little-endian) — the CD-quality baseline that every editor reads. Switch to pcm_s24le for 24-bit mastering depth, pcm_s32le for 32-bit headroom, or pcm_alaw / pcm_mulaw for legacy telephony workflows (8-bit logarithmic, ~64 kbit/s).WebM is Google's open container (released May 2010) that wraps VP8/VP9/AV1 video and Vorbis or Opus audio. WAV is the Microsoft and IBM RIFF container from 1991 that stores uncompressed linear PCM — the lingua franca of audio editing. Converting from WebM to WAV decodes the lossy Opus/Vorbis stream back to raw samples so DAWs, transcription engines, and broadcast tools can work with it natively.
| Property | WebM | WAV |
|---|---|---|
| First released | 2010 (Google / On2 / Xiph / Matroska) | 1991 (Microsoft & IBM) |
| Container | Matroska-based | RIFF |
| Typical contents | VP8/VP9/AV1 video + Vorbis or Opus audio | Uncompressed LPCM (also A-law, μ-law, ADPCM) |
| Compression | Lossy audio (Opus/Vorbis) | Usually uncompressed (lossless) |
| File size (1 min stereo, 44.1 kHz) | ~0.5–1 MB at 128 kbps Opus | ~10 MB at 16-bit / ~15 MB at 24-bit |
| Max file size | Effectively unbounded (Matroska 64-bit) | ~4 GiB (32-bit RIFF header — about 6.8 h of 16-bit / 44.1 kHz stereo) |
| Browser playback | Native in Chrome, Firefox, Edge, Opera; Safari 16+ desktop / 17.4+ iOS | Native in all major browsers |
| Editor support | Limited — many DAWs need a re-wrap | Universal across DAWs, broadcast, transcription |
| Best for | Streaming and web delivery | Editing masters, archiving, ASR input |
| Setting | Choose when | Notes |
|---|---|---|
pcm_s16le @ 44.1 kHz stereo |
General editing, music, podcast masters | CD-quality baseline; ~10 MB per minute |
pcm_s16le @ 16 kHz mono |
Whisper / Deepgram / Google ASR | Smallest file that still meets ASR specs; ~1.9 MB per minute |
pcm_s24le @ 48 kHz stereo |
Video-post handoff, mastering | Broadcast standard; ~17 MB per minute |
pcm_s32le @ 48 kHz stereo |
Mixing headroom, intermediate renders | Avoids clipping during heavy processing |
pcm_alaw / pcm_mulaw @ 8 kHz mono |
Legacy telephony, IVR systems | 8-bit logarithmic; G.711 compatible |
WebM audio is lossy and compressed (Opus typically targets 64–128 kbps for stereo speech and music). WAV is uncompressed PCM, so a 1-minute 44.1 kHz / 16-bit stereo clip is roughly 10 MB regardless of source bitrate. Converting back to WAV doesn't restore quality lost in the original Opus encode — it just unpacks the compressed samples into raw form so editors can read them.
Both. WebM officially supports Vorbis (the original 2010 spec) and Opus (added in 2013). The decoder auto-detects which codec the file contains and produces a PCM WAV either way. If your file has multiple audio tracks, the default/first track is used.
Match the source if you can. WebM files recorded by browser MediaRecorder or WebRTC typically use 48 kHz Opus, so 48 kHz keeps the conversion sample-accurate. CDs and most music libraries use 44.1 kHz. Picking the "wrong" rate forces a resample that adds minor filtering artifacts but is usually inaudible — only pick 8/16/24 kHz if you specifically need a smaller file for speech or telephony.
For speech, music distribution, or anything destined for streaming, 16-bit is fine and is what most DAWs default to. Pick 24-bit when handing off to a video editor or mastering engineer who needs headroom for further processing. 32-bit float is overkill for a source that was already lossy WebM — it doesn't add real precision, just larger files.
This page always writes PCM WAV. If you want to keep the original Opus or Vorbis stream without a decode step (smaller file, identical quality), use the WebM to MP3 or a direct demux tool instead. WAV by definition is a PCM container, so a "WebM-to-WAV" conversion always involves decoding to raw samples.
WAV's RIFF header uses a 32-bit unsigned integer for file size, capping output at about 4 GiB — roughly 6.8 hours of 16-bit / 44.1 kHz stereo, or 3.4 hours of 24-bit / 48 kHz stereo. For longer recordings (multi-hour lectures, all-day streams), trim the WebM first using the Audio Trim controls or split the source.
Files written by MediaRecorder (Chrome screen recording, in-browser voice memos, WebRTC captures) often omit the duration field in the Matroska header — the file plays fine but seek bars and FFmpeg's -t flag get confused. The conversion still works because the decoder walks the whole stream; if downstream tools need a correct duration, the freshly written WAV will have an accurate one.
Yes. Use the Trim control to set a start time and duration so only the segment you want gets written to WAV. For more elaborate edits (multiple regions, fades, normalization), bounce the full file to WAV first, then open it in the Audio Cutter or any DAW.
The Conversion runs on our servers and your file is deleted shortly after — useful for confidential interviews, internal meeting recordings, or unreleased music. Compare with WAV to MP3 if you later need to compress for sharing, or WebM to MP4 if you actually wanted to keep the video.