WMV to WAV Converter

How to Convert WMV to WAV Online

1. Upload Your WMV File: Drag and drop or click "+ Add Files" to add one or more .wmv clips. Batch is supported, and processing happens in your browser session — no account, no watermark, no email.
2. Pick Audio Sample Rate and Channel: The defaults keep the WMV source's original rate and channel layout, which is the safest pick when you just want a clean PCM copy. For editing/mastering, set Audio Sample Rate to 48000 Hz (broadcast/video-post standard) or 44100 Hz (CD/streaming master), and Audio Channel to Stereo or Mono. The output codec defaults to PCM signed 16-bit little-endian (pcm_s16le) — the standard WAV payload for compatibility with every DAW.
3. Trim to the Section You Need (Optional): Use the Trim control to crop to a start time and duration in hours/minutes/seconds/milliseconds. Useful when you only need a quote, a music cue, or a single sentence from a longer screen recording or lecture capture.
4. Convert and Download: Click Convert. The encoder strips the WMV video track, decodes the WMA audio stream from the ASF container, and writes a RIFF/WAV file with linear PCM samples. Download individually or grab the zipped batch.

Why Convert WMV to WAV?

WMV is a Microsoft video container (.wmv, wrapping an ASF stream introduced with WMV 7 in 1999) that almost always carries a Windows Media Audio (WMA) track. WMA is lossy. The moment you need to edit the audio — re-cut a voiceover, isolate music for licensing, run a transcript through Whisper, or master a podcast episode — you want uncompressed linear PCM in a WAV (RIFF) container. WAV was published by Microsoft and IBM in August 1991 and is the lingua franca of every DAW, broadcast workflow, and speech-to-text engine on the planet.

• Editing master for DAWs — Pro Tools, Logic Pro, Ableton Live, Reaper, Audacity, and Adobe Audition all import WAV natively and prefer 24-bit/48 kHz PCM as a working format. Importing WMA forces an extra decode pass and locks you to whatever quality the original WMA encoder chose.
• Voiceover and podcast post — narrators record VO over Zoom/Teams (which often saves WMV/ASF screen recordings on Windows). Convert the captured WMV to WAV so you can EQ, de-ess, level, and noise-reduce without re-compressing a lossy source twice.
• Speech-to-text and transcription — OpenAI Whisper, Deepgram, AWS Transcribe, and Google Speech-to-Text all accept WAV and tend to produce cleaner output from 16 kHz mono PCM than from compressed WMA. Set Audio Sample Rate to 16000 Hz and Channel to Mono before running ASR.
• Archival and preservation — the U.S. Library of Congress lists WAV/PCM as a preferred format for digital audio preservation; WMA is not. If you're archiving lectures, oral histories, or interviews recorded on a Windows machine, WAV is the long-term format.
• Broadcast and video editing — Avid Media Composer, DaVinci Resolve, and Premiere Pro all consume 48 kHz/24-bit WAV without transcoding. Strip the audio off old WMV training videos or webinars and drop the WAV directly onto a timeline.
• Sampling and remixing — producers pull dialogue, sound effects, or music cues from old WMV captures (game footage, TV rips, screen recordings). A WAV lets you time-stretch, pitch-shift, and resample without compounding lossy artifacts.

WMV (ASF/WMA) vs WAV (RIFF/PCM)

Sample Rate and Bit Depth Guide

Each extra bit of depth adds roughly 6 dB of dynamic range, so 16-bit gives ~96 dB and 24-bit gives ~144 dB. The audible difference at sane listening levels is negligible on playback, but the headroom matters when you process: EQ, compression, time-stretch, and pitch-shift all chew into the noise floor. Master at 24-bit, deliver at 16-bit with dither.

Frequently Asked Questions

Why does the output WAV come out 10–20 times larger than the source WMV?

Because WMV's audio stream is lossy WMA at roughly 64–192 kbps, while WAV/PCM is uncompressed. One minute of stereo 44.1 kHz 16-bit PCM is ~10 MB; the same minute as 128 kbps WMA is ~1 MB. The size jump is unavoidable when going from a compressed codec to raw PCM — that's exactly what makes WAV editable without further generation loss. Convert to WAV here for the editing pass, then export as WAV to MP3 or WAV to FLAC for delivery.

Will I gain audio quality by converting WMA-inside-WMV to WAV?

No. Converting lossy audio to a lossless container does not recover detail that was thrown away during the WMA encode. What you gain is editability: every subsequent edit, EQ pass, or processing step happens in the lossless PCM domain instead of re-decoding and (sometimes) re-encoding the WMA. Think of it as freezing the audio at its current quality so you can work on it without further degradation.

Which bit depth should I pick — 16, 24, or 32?

Pick 24-bit (pcm_s24le) if you'll edit, mix, or master the audio — the extra 48 dB of dynamic range gives you headroom for processing without lifting the noise floor. Pick 16-bit (pcm_s16le, the default) if the file is for distribution, archival of a speech recording, or any consumer playback context. 32-bit signed (pcm_s32le) is overkill for almost every WMV source since the WMA audio it was decoded from rarely justifies that resolution.

What sample rate should I choose for transcription?

16000 Hz mono. Whisper, Deepgram, AWS Transcribe, and Google's Speech-to-Text API all downsample to 16 kHz internally, so providing 48 kHz stereo just wastes bandwidth and storage. Setting Audio Sample Rate to 16000 Hz and Audio Channel to Mono produces the leanest WAV that ASR engines actually consume — typically ~30 MB per hour instead of ~600 MB.

My WMV has multiple audio tracks (dub + original). Which one ends up in the WAV?

The default behavior extracts the first/primary audio stream declared in the ASF header — usually the language the file was authored in. ASF supports multiple audio streams, but most consumer WMV files only have one. If your file has alternates (multi-language DVDs ripped to WMV, dual-mono ad masters), you may need a desktop tool like ffmpeg with -map 0:a:N to pick a specific stream.

Why does the WAV play but my DAW says "incompatible format"?

A handful of DAWs (older Pro Tools versions, some embedded recorders) only accept signed 16-bit PCM little-endian and reject 24-bit, 32-bit, or float WAVs. If your DAW chokes, re-run the conversion with the default codec (pcm_s16le) and a sample rate matching the session (44100 or 48000 Hz). That's the most universally accepted WAV payload.

Can I keep the video too, just in case?

This tool extracts audio only — the video track is discarded because the output is a WAV (audio container). If you want to keep the video and just switch the wrapper, use WMV to MP4 instead. If you want both — edit the audio separately and re-mux later — convert to WAV here for the audio edit, then use a video editor to swap the audio track back onto the original WMV/MP4.

Does WAV support metadata like artist, title, and album?

Yes, via the LIST/INFO chunk in the RIFF container (and increasingly via embedded ID3 chunks). Most DAWs and players read these tags, though the ecosystem is less standardized than MP3's ID3 or FLAC's Vorbis comments. If you need rich metadata for a music library, consider exporting WAV to FLAC — same lossless quality, smaller files, better tag support.

Is there a file-size limit I should know about?

The WAV format itself caps at roughly 4 GiB because the RIFF header stores file size as a 32-bit unsigned integer. At 44.1 kHz/16-bit stereo, that's about 6.8 hours of audio. For longer recordings you'd need RF64 or Wave64 extensions, which most consumer tools don't write. In practice, if your source WMV is over ~4 hours of speech, split it before converting or use Audio Cutter on the resulting WAV.