Extract Audio from WMV to WAV — Free Online Tool
Extract the audio track from a WMV file and save it as an uncompressed WAV — converting Windows Media Audio (WMA/wmav2) into raw PCM audio (pcm_s16le) that any audio editor, DAW, or broadcast system can work with. Ideal when you need a lossless-quality audio file from a WMV source without any further encoding artifacts.
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
WMV files store audio using the wmav2 codec (Windows Media Audio 2), which is a lossy compressed format — similar in concept to MP3 but proprietary to Microsoft. During this conversion, FFmpeg reads the ASF container (the underlying format of WMV), discards the video stream entirely, and decodes the wmav2 audio to raw PCM samples. Those samples are then written into a WAV container using the pcm_s16le codec — 16-bit signed little-endian PCM — which is completely uncompressed. This means the output WAV will be significantly larger than the WMV source, but it introduces no additional lossy encoding step. The audio quality is bounded by the original wmav2 encoding in the WMV; you cannot recover information lost when the WMV was first created, but you avoid any further degradation.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In the browser-based tool, this runs via FFmpeg.wasm compiled to WebAssembly — no server involved. On your desktop, this calls your locally installed FFmpeg binary. |
-i input.wmv
|
Specifies the input file — your WMV source. FFmpeg reads the ASF container, identifies the wmav2 audio stream (and any video streams), and makes them available for processing. |
-vn
|
Stands for 'video none' — instructs FFmpeg to discard the video stream entirely. Since the goal is audio-only WAV output, this prevents FFmpeg from attempting to encode or include any video data, which the WAV container cannot hold anyway. |
-c:a pcm_s16le
|
Sets the audio codec to pcm_s16le — 16-bit signed little-endian PCM. This decodes the compressed wmav2 audio from the WMV and writes it as fully uncompressed raw samples in the WAV file, the format expected by virtually all audio editors, DAWs, and broadcast tools. |
output.wav
|
The output filename, which also tells FFmpeg to use the WAV container format. The resulting file will be an uncompressed WAV containing the decoded audio from the WMV source, ready for editing, archiving, or use in any audio pipeline. |
Common Use Cases
- Import WMV training videos or lecture recordings into a DAW like Audacity or Adobe Audition, which work natively with WAV but often struggle with wmav2-encoded WMA audio inside WMV containers.
- Prepare audio from Windows Media Player recordings or screen captures for broadcast or podcast production pipelines that require uncompressed WAV input.
- Extract narration or dialogue from corporate WMV presentations to use in video re-edits, where you need a clean, editable audio file free of any additional lossy compression.
- Archiving the audio content of legacy WMV files in an open, uncompressed format that will remain readable without proprietary Windows Media codecs decades from now.
- Feed WMV-sourced audio into automated transcription services or speech recognition tools that require standard PCM WAV input rather than compressed Windows Media Audio.
- Strip the soundtrack from a WMV gameplay recording or screen capture to synchronize or mix it with footage edited in a non-linear video editor that doesn't support ASF/WMV natively.
Frequently Asked Questions
Yes, within the limits of the original encoding. The wmav2 audio in the WMV is decoded to PCM and written as-is into the WAV — no additional lossy compression step occurs. However, the WMV's audio was already compressed with wmav2 when the file was created, so any quality loss from that original encoding is permanent and will be present in the WAV output. The WAV simply preserves what was there; it does not restore detail that wmav2 discarded.
WMV stores audio using wmav2, a lossy codec that compresses audio data significantly — typically at 128 kbps or less. WAV with pcm_s16le stores every audio sample as raw, uncompressed 16-bit data, which for stereo audio at 44.1 kHz works out to about 10 MB per minute. A 10-minute WMV with a small audio track might produce a WAV that is 100 MB or more. This size increase is expected and is the nature of uncompressed audio.
WMV files can carry metadata in the ASF container (title, author, copyright, etc.), but the WAV format has very limited and inconsistently supported metadata fields. FFmpeg may carry over some basic tags into the WAV's INFO chunk, but most WMV metadata — especially Windows Media-specific DRM or licensing information — will not be present in the output. If metadata preservation is critical, check the output file in a tag editor after conversion.
Yes. The command uses '-c:a pcm_s16le' to produce 16-bit WAV. To get 24-bit output, replace that with '-c:a pcm_s24le', giving you 'ffmpeg -i input.wmv -vn -c:a pcm_s24le output.wav'. For 32-bit float PCM, use 'pcm_f32le'. Keep in mind that the source audio is wmav2-compressed, so increasing the bit depth won't recover lost quality — it just gives downstream tools a higher-headroom working format, which can be useful in professional audio workflows.
On Windows, open Command Prompt and run: 'for %f in (*.wmv) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav"'. On macOS or Linux, use: 'for f in *.wmv; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.wmv}.wav"; done'. Both loops apply the same extraction command to every WMV in the current directory, naming each output WAV after the original file. The browser-based tool processes one file at a time, so the FFmpeg command is the best path for bulk jobs.
No. WMV supports Microsoft's DRM (Digital Rights Management), which encrypts the media content to prevent unauthorized playback and extraction. FFmpeg cannot decrypt DRM-protected WMV files, and neither can the browser-based tool. If FFmpeg reports an error mentioning encryption or DRM when processing your file, the file is protected and you would need to use licensed playback software or obtain an unprotected version of the content.
Technical Notes
WMV is built on the ASF (Advanced Systems Format) container, which is why FFmpeg sometimes requires the '-f asf' flag in edge cases — though for audio extraction it typically auto-detects the format from the file header. The default audio codec in WMV is wmav2 (Windows Media Audio 2), a transform-based lossy codec with characteristics broadly similar to MP3 but using a proprietary algorithm. When decoded to pcm_s16le for WAV output, the audio is represented as 16-bit signed little-endian PCM samples — the standard format used by audio CDs and most professional audio tools. WAV supports multiple PCM variants (8-bit, 16-bit, 24-bit, 32-bit float) and even compressed codecs like ADPCM, but pcm_s16le is the safest choice for maximum compatibility across DAWs, broadcast systems, and consumer software. One known limitation: WMV supports multiple audio tracks, but the WAV format does not. By default, FFmpeg will extract only the first audio track (stream 0:a:0). If your WMV contains multiple audio streams — for example, different language dubs — you can target a specific stream by adding '-map 0:a:1' (for the second audio track) to the command. Also note that WAV has a theoretical 4 GB file size limit due to its 32-bit chunk size header; for very long audio at high sample rates, consider using the W64 or RF64 format variants instead.