Convert WMV to AAC — Free Online Tool
Extract and convert the audio track from a WMV file into a standalone AAC file, stripping away the ASF container and WMA audio codec in favor of the modern, widely-supported AAC format. AAC delivers better sound quality than WMA at equivalent bit rates and is natively compatible with iOS, Android, web browsers, and iTunes.
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 WMA v2 codec (wmav2) inside Microsoft's proprietary ASF (Advanced Systems Format) container. During this conversion, FFmpeg discards the video stream entirely and decodes the wmav2 audio track from the ASF container, then re-encodes it into AAC using the native FFmpeg AAC encoder. The result is a standalone .aac audio file — a raw AAC stream without a container wrapper. Because both WMA and AAC are lossy formats, this is a lossy-to-lossy transcode, meaning there will be a small, typically inaudible quality loss compared to the original WMA audio. The ASF-specific metadata and any DRM licensing embedded in the WMV file are not carried over to the AAC output.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program. In this browser tool, this runs via FFmpeg.wasm entirely in your browser without any server upload; when run locally, it calls your desktop FFmpeg installation. |
-i input.wmv
|
Specifies the input WMV file. FFmpeg reads the ASF container and identifies the available streams — typically an msmpeg4 video track and a wmav2 audio track — from which it will extract only the audio for this conversion. |
-c:a aac
|
Sets the audio codec to AAC using FFmpeg's built-in native AAC encoder. This decodes the WMA v2 audio from the WMV file and re-encodes it as AAC, which is natively supported by iOS, Android, web browsers, and iTunes without any plugins. |
-b:a 128k
|
Sets the AAC audio output bit rate to 128 kilobits per second, which is the standard default for AAC and produces transparent or near-transparent quality for most speech and music content when converting from a WMA source at similar quality. |
output.aac
|
Defines the output filename and format. The .aac extension tells FFmpeg to write a raw AAC bitstream. Change this to output.m4a if you need AAC wrapped in an MPEG-4 container for better iTunes compatibility and metadata support. |
Common Use Cases
- Extracting the audio commentary or narration from a WMV screen recording or presentation to use in a podcast or audio-only distribution
- Converting Windows Media Player-locked WMV audio to AAC so it can be imported into GarageBand, Logic Pro, or other Apple ecosystem software
- Pulling the audio track from a legacy WMV corporate training video so it can be played on iOS or Android mobile devices without needing a WMV-compatible app
- Stripping the audio from a WMV file to create an AAC version suitable for embedding in an HTML5 web page using the <audio> tag
- Archiving the audio content of old WMV home videos or recordings into AAC, which has broader long-term platform support than the proprietary WMA codec
- Preparing audio extracted from a WMV file for upload to iTunes, Apple Podcasts, or other Apple platforms that require or prefer AAC encoding
Frequently Asked Questions
Yes, there will be some quality loss because this is a transcode between two lossy formats — WMA v2 in the WMV file gets decoded and then re-encoded as AAC. Each generation of lossy encoding introduces additional compression artifacts. However, at the default 128k bit rate, the difference is typically inaudible for speech and difficult to detect even for music. If quality is critical, increase the output bit rate to 192k or 256k to minimize the impact of re-encoding.
The video stream is completely discarded. FFmpeg reads the WMV file, ignores the msmpeg4 or msmpeg4v2 video track, and only processes the audio. The output is a pure AAC audio file with no video data. If you need to keep the video, you would need to convert to a format like MP4 instead of AAC.
No. WMV files protected with Microsoft's DRM (Digital Rights Management) cannot be decoded by FFmpeg or FFmpeg.wasm. The conversion will fail or produce silent/corrupt output. This tool only works with unprotected WMV files. DRM-protected WMV files typically require Windows Media Player or authorized Microsoft DRM licensing software to play back.
Change the value after the -b:a flag to adjust the audio bit rate. For example, use -b:a 192k or -b:a 256k for higher quality output, or -b:a 96k for smaller file sizes at the cost of some audio fidelity. The AAC format supports bit rates from 64k up to 320k. For music, 192k is a good balance; for voice or speech, 96k or 128k is usually sufficient.
The single-file command shown won't batch process on its own, but on Linux or macOS you can wrap it in a shell loop: for f in *.wmv; do ffmpeg -i "$f" -c:a aac -b:a 128k "${f%.wmv}.aac"; done. On Windows Command Prompt you can use: for %f in (*.wmv) do ffmpeg -i "%f" -c:a aac -b:a 128k "%~nf.aac". This is particularly useful for the desktop FFmpeg command since the browser tool processes one file at a time.
The command outputs a raw AAC bitstream (.aac) without wrapping it in a container. If you need an .m4a file — which is AAC audio inside an MPEG-4 container, preferred by iTunes and Apple devices — change the output filename in the FFmpeg command to output.m4a. FFmpeg will automatically use the MP4 container, which also preserves metadata like track title and artist more reliably than a raw AAC stream.
Technical Notes
WMV files use Microsoft's ASF (Advanced Systems Format) container, which requires the -f asf flag when FFmpeg is writing ASF output — however, since we are reading WMV as input and writing AAC as output, no special input flags are needed. The audio codec in most WMV files is WMA v2 (wmav2), a proprietary Microsoft codec. FFmpeg can decode wmav2 freely but re-encodes to AAC using its built-in native AAC encoder, which produces good quality output. The higher-quality libfdk_aac encoder is an alternative if you compile FFmpeg with that library, achievable by replacing -c:a aac with -c:a libfdk_aac in the command. Metadata embedded in the ASF container (such as title, artist, or copyright fields) is generally not carried over to a raw .aac output; switching the output to .m4a resolves this. WMV files that contain multiple audio tracks will only have the first (default) audio track extracted by this command — to select a specific track, add -map 0:a:1 (or the appropriate index) to the command before the output filename.