Convert WMV to WAV — Free Online Tool
Convert WMV video files to WAV audio by extracting the WMA-encoded audio stream and decoding it to uncompressed 16-bit PCM — the standard lossless audio format used in professional and broadcast workflows. This tool runs entirely in your browser using FFmpeg.wasm, so no files leave your device.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your WMV file here
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Free — no uploads, no signups. Your files never leave your browser.
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
WMV files store audio using the Windows Media Audio v2 (wmav2) codec, a proprietary lossy compression format developed by Microsoft. During this conversion, FFmpeg demuxes the WMV container (technically an ASF file), discards the video stream entirely, and decodes the wmav2 audio data. The decoded audio is then re-encoded using the pcm_s16le codec — signed 16-bit little-endian PCM — and written into a WAV container. Because wmav2 is a lossy codec, the original audio signal was already compressed when the WMV was created; converting to WAV produces a lossless PCM file, but it cannot recover detail lost during the original WMA encoding. The result is a high-compatibility, uncompressed WAV file at whatever fidelity the WMV source preserved.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In this browser-based tool, FFmpeg runs as a WebAssembly (wasm) module locally in your browser — no server is involved. When running locally on your desktop, this calls your system-installed FFmpeg executable. |
-i input.wmv
|
Specifies the input file — a WMV video using the ASF container, typically carrying a wmav2 audio stream and an msmpeg4 video stream. FFmpeg automatically detects the ASF format and available streams without needing an explicit format flag on input. |
-c:a pcm_s16le
|
Sets the audio codec for the output to signed 16-bit little-endian PCM — the standard uncompressed audio encoding used in WAV files. This decodes the lossy wmav2 audio from the WMV and writes it as raw PCM samples, producing a lossless WAV that is universally compatible with audio editors, broadcast systems, and transcription tools. |
output.wav
|
Defines the output filename and triggers FFmpeg to use the WAV container format. WAV is an audio-only container, so the video stream from the WMV is automatically discarded. The resulting file contains only the uncompressed PCM audio track. |
Common Use Cases
- Extracting voice-over or narration audio from a WMV screen recording or presentation to edit in a DAW like Audacity or Adobe Audition
- Pulling the audio track from a Windows Media Player-era video archive to use as source material for a podcast or audio project
- Preparing audio from a WMV training video for transcription services that require standard WAV input rather than proprietary WMA or ASF formats
- Converting WMV interview recordings to WAV for import into professional video editing software like DaVinci Resolve or Premiere Pro, which handle PCM WAV more reliably than wmav2
- Extracting audio from legacy corporate WMV files for archival storage in an open, uncompressed format that does not depend on Microsoft codec support
- Separating the audio from a WMV music video to create a clean WAV file for mixing or mastering workflows
Frequently Asked Questions
No — the audio quality is capped at whatever fidelity the original WMV preserved. WMV files typically store audio using the wmav2 codec, which is a lossy format similar to MP3. When FFmpeg decodes that stream to PCM WAV, the output is an exact uncompressed representation of the already-lossy source. The WAV file will be larger and technically lossless going forward, but the detail lost during the original WMA encoding cannot be restored.
The WAV format stores audio as raw, uncompressed 16-bit PCM samples rather than compressed WMA data. A typical stereo 44.1kHz PCM WAV uses roughly 10 MB per minute, whereas the same audio in wmav2 at 128k might occupy under 1 MB per minute. The size increase is entirely expected and does not indicate a quality improvement — it simply reflects the absence of compression in the output format.
Yes. FFmpeg reads the sample rate and channel count from the wmav2 stream in the WMV and writes those parameters into the WAV file without altering them. If the WMV audio was recorded at 44.1kHz stereo, the output WAV will also be 44.1kHz stereo. If you need a specific sample rate for a target application, you can add '-ar 48000' to the command to resample the output.
Yes. The default command uses '-c:a pcm_s16le' for 16-bit PCM, which is the most universally compatible WAV format. To produce a 24-bit WAV, replace that flag with '-c:a pcm_s24le'. For 32-bit float PCM — common in professional audio workflows — use '-c:a pcm_f32le'. Higher bit depths produce larger files but may be preferred when the output will be further processed in a DAW.
On Linux or macOS, you can use a shell loop: 'for f in *.wmv; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.wmv}.wav"; done'. On Windows Command Prompt, use: 'for %f in (*.wmv) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav"'. This processes each WMV file in the current directory and writes a matching WAV file alongside it. Files over 1GB are best handled this way locally rather than through the browser tool.
The video stream is silently discarded. WAV is a pure audio container and has no capacity to store video data. FFmpeg automatically drops the video when the output format is WAV — you do not need to add a '-vn' flag explicitly, though including it is harmless. Only the first audio track from the WMV will be written to the WAV, since WAV does not support multiple audio tracks.
Technical Notes
WMV files use the Advanced Systems Format (ASF) container, which is why the FFmpeg command for writing WMV requires the '-f asf' flag — though for reading, FFmpeg detects the format automatically. The audio codec in most WMV files is wmav2, a Microsoft proprietary lossy codec; less commonly you may encounter AAC or even MP3 audio inside a WMV container. In all cases, FFmpeg will decode whatever audio codec is present and re-encode to pcm_s16le. The default output codec pcm_s16le (signed 16-bit little-endian PCM) is the standard encoding for CD-quality audio and is accepted by virtually every piece of audio software, broadcast system, and transcription service. WMV files do not carry embedded subtitle or chapter metadata, so there is nothing to lose on that front. However, any ID3-style or ASF metadata tags (title, artist, album) stored in the WMV will not automatically transfer to the WAV file, since WAV has limited and non-standardized metadata support. If metadata preservation matters, consider using FLAC as an output format instead, as it supports both lossless PCM storage and robust metadata tagging.