Extract Audio from WMV to MP3 — Free Online Tool
Extract audio from WMV files and save it as MP3 by discarding the video stream and re-encoding the WMV's default wmav2 audio track using the LAME MP3 encoder. The resulting MP3 is universally compatible across devices and platforms that the original Windows Media Audio codec never was.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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 typically carry audio encoded in Microsoft's proprietary wmav2 (Windows Media Audio v2) codec, wrapped in an ASF (Advanced Systems Format) container. Because MP3 is a pure audio format using the entirely different libmp3lame codec, a straightforward stream copy is not possible — the audio must be fully decoded from wmav2 and then re-encoded as MPEG Audio Layer III. The video stream is discarded entirely using the -vn flag, so FFmpeg never decodes or processes the video data, keeping conversion fast. The re-encoding step does introduce a second generation of lossy compression: the audio was already lossy in WMV, and encoding to MP3 applies lossy compression again. At 128k bitrate the result is generally transparent for speech and acceptable for music, but for critical listening you may want to raise the bitrate to 192k or 320k.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser version of this tool, this runs as FFmpeg.wasm compiled to WebAssembly — no files leave your device. |
-i input.wmv
|
Specifies the input WMV file. FFmpeg automatically detects the ASF container and identifies the wmav2 audio and video streams inside without needing additional format hints on input. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the WMV's video stream and produce an audio-only output. This is what makes the result an MP3 rather than a video file, and it avoids wasting time decoding video frames. |
-c:a libmp3lame
|
Sets the audio codec to libmp3lame, the open-source LAME encoder for MPEG Audio Layer III. This is required because the WMV's wmav2 audio cannot be stream-copied into an MP3 file — it must be fully decoded and re-encoded as MP3. |
-b:a 128k
|
Sets the MP3 output bitrate to 128 kilobits per second, a widely accepted baseline for perceptually transparent audio at manageable file sizes. You can raise this to 192k or 320k for higher fidelity, especially if the source WMV was encoded at a high audio bitrate. |
output.mp3
|
Defines the output filename and, by its .mp3 extension, tells FFmpeg to write a raw MPEG Audio Layer III bitstream — the standard MP3 format supported by every modern device, browser, and media player. |
Common Use Cases
- Pull the audio commentary track from a WMV screen recording or software tutorial to repurpose it as a podcast episode or voiceover asset
- Extract music or sound effects from old Windows Media Player-era WMV files archived on a hard drive, making them playable on modern devices that no longer ship with wmav2 decoders
- Convert WMV recordings of webinars, conference talks, or lectures into MP3 files for offline listening on a phone or MP3 player
- Strip the audio from a WMV video received from a Windows-based client or colleague to edit it in audio software like Audacity or GarageBand, which may not support WMV natively
- Reduce file size dramatically by discarding the video stream from large WMV recordings where only the spoken audio content is needed for transcription or archiving
- Prepare WMV-sourced audio for upload to podcast platforms, audiobook distributors, or streaming services that require MP3 input and do not accept wmav2 or ASF containers
Frequently Asked Questions
Yes, some quality loss is unavoidable. WMV's default wmav2 audio is already lossy, and encoding it to MP3 with libmp3lame applies a second round of lossy compression. In practice, at 128k or above the degradation is subtle and often imperceptible for speech, but critical music listeners may notice slight artifacts. To minimize quality loss, use the highest bitrate option (320k) when the source WMV was encoded at a high bitrate. There is no way to recover quality that was already discarded by the original wmav2 encoding.
Stream copying would only work if the audio in the WMV were already encoded as MP3 (libmp3lame), which is possible but uncommon — most WMV files use Microsoft's own wmav2 codec. Since MP3 is a raw audio bitstream format and not a container, it cannot simply wrap a wmav2 stream. The audio must be fully decoded and re-encoded to libmp3lame, making re-encoding mandatory for the typical WMV-to-MP3 conversion.
Replace the value after -b:a in the command. For example, to export at 320k for higher quality, use: ffmpeg -i input.wmv -vn -c:a libmp3lame -b:a 320k output.mp3. For a smaller file at the cost of quality, try 96k or 64k. The default of 128k is a reasonable balance for speech-heavy content. This tool also lets you select the bitrate from a dropdown before running the conversion in your browser.
FFmpeg will attempt to copy compatible metadata fields — such as title, artist, and album — from the ASF container's metadata into the MP3 file's ID3 tags. However, ASF and ID3 use different metadata schemas, so not every field maps cleanly, and some WMV-specific metadata (like DRM licensing information or Windows Media Player ratings) will be dropped entirely. You may want to inspect and clean the ID3 tags in the output MP3 using a tag editor afterward.
Yes, on the command line you can use a shell loop. On Linux or macOS: for f in *.wmv; do ffmpeg -i "$f" -vn -c:a libmp3lame -b:a 128k "${f%.wmv}.mp3"; done. On Windows Command Prompt: for %f in (*.wmv) do ffmpeg -i "%f" -vn -c:a libmp3lame -b:a 128k "%~nf.mp3". The browser-based tool on this page processes one file at a time, so the command-line approach is recommended for batch jobs.
FFmpeg cannot process DRM-protected WMV files. The ASF container supports Microsoft's DRM (Digital Rights Management) system, and if the file is protected, FFmpeg will output an error and produce no output. This tool and the FFmpeg command work only on unprotected WMV files. DRM-protected WMV files must be handled through licensed playback software, and circumventing DRM may be restricted under local law.
Technical Notes
WMV files use the ASF (Advanced Systems Format) container, which FFmpeg addresses with the -f asf flag when writing but reads automatically on input. The most common audio codec in WMV is wmav2 (Windows Media Audio v2), a proprietary Microsoft codec offering quality comparable to MP3 at similar bitrates but with very narrow native decoder support outside of Windows environments. The libmp3lame encoder used here is the gold-standard open-source MPEG Layer III implementation, producing MP3 files with broad ID3v2 tag support and compatibility with virtually every audio player, browser, smart speaker, and streaming platform. One important limitation: MP3 does not support multiple audio tracks — if your WMV contains multiple audio streams (e.g., a commentary track alongside a main audio track), FFmpeg will select only the first audio stream by default. To extract a non-default track, add -map 0:a:1 (for the second audio track) to the command. Also note that WMV's video stream is not touched at all during this operation; the -vn flag causes FFmpeg to skip video decoding entirely, which keeps processing time short even for long WMV files.