Convert WAV to WMA — Free Online Tool
Convert WAV audio files to WMA format using the WMA v2 codec (wmav2) directly in your browser — no upload required. This conversion transcodes uncompressed PCM audio from WAV into Microsoft's proprietary Windows Media Audio lossy format, making files significantly smaller while retaining good perceptual quality at 128k bitrate by default.
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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
WAV files typically store audio as uncompressed PCM data (most commonly 16-bit signed little-endian, pcm_s16le), which means every audio sample is stored verbatim with no compression — resulting in large file sizes. During conversion to WMA, FFmpeg re-encodes this raw PCM audio using the wmav2 codec, which applies Microsoft's proprietary perceptual audio compression algorithm. The codec analyzes the audio signal and discards frequency information that is less perceptible to human hearing, similar in philosophy to MP3 but using Microsoft's own encoding model. This is a full re-encode, not a remux — the audio data is transformed from lossless PCM into lossy WMA, which cannot be reversed without quality loss. At the default 128k bitrate, file sizes are typically reduced by 80–90% compared to a 16-bit stereo WAV at CD quality (1411 kbps).
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles all decoding, encoding, and format conversion. In this browser-based tool, FFmpeg runs locally via WebAssembly (ffmpeg.wasm), so no data leaves your machine. |
-i input.wav
|
Specifies the input file — your WAV audio file. FFmpeg will automatically detect the WAV container and decode whichever internal PCM codec the file uses (commonly pcm_s16le for 16-bit CD-quality audio) into raw audio samples for re-encoding. |
-c:a wmav2
|
Sets the audio codec to wmav2 (Windows Media Audio v2), Microsoft's proprietary lossy audio compression algorithm. This is the modern and higher-quality WMA codec, preferred over the older wmav1, and is the standard codec expected inside .wma files on Windows platforms. |
-b:a 128k
|
Sets the audio bitrate to 128 kilobits per second for the WMA output. This controls the trade-off between file size and audio quality — 128k is a widely accepted default that balances perceptual quality with significant file size reduction compared to the original uncompressed WAV. Increase this value (e.g., 192k or 256k) for better fidelity on music content. |
output.wma
|
Defines the output filename and tells FFmpeg to wrap the wmav2-encoded audio in an ASF container with the .wma extension. The .wma extension is essential — it signals to Windows and media players that this is a Windows Media Audio file using the ASF format. |
Common Use Cases
- Preparing audio files for playback or distribution on Windows-centric platforms and legacy Windows Media Player libraries that expect WMA format natively.
- Reducing the storage footprint of large WAV recordings — such as multi-hour voice recordings or interview archives — for archiving on Windows devices or OneDrive with format compatibility in mind.
- Delivering audio assets to clients or systems that require WMA, such as certain corporate e-learning authoring tools (like older versions of Articulate or Lectora) that accept WMA but not WAV.
- Converting studio-recorded WAV files to WMA for distribution on older portable media players or in-car audio systems that support WMA but lack storage for uncompressed WAV files.
- Encoding broadcast voiceovers or narration captured as high-quality WAV into WMA for embedding in Windows-based multimedia presentations or legacy Microsoft PowerPoint files.
- Producing smaller WMA files from WAV masters for streaming over low-bandwidth internal networks or intranets where WMA's streaming-optimized container offers compatibility advantages.
Frequently Asked Questions
Yes — this conversion is lossy, meaning audio quality cannot be fully recovered once encoded to WMA. WAV with pcm_s16le stores audio at full CD quality (typically 1411 kbps for stereo), while the default 128k WMA output is compressed to roughly 9% of that data rate. At 128k, wmav2 generally produces audio that sounds acceptable for voice, podcasts, and casual music listening, but critical listeners may notice artifacts on complex music or high-frequency content. If quality is a priority, consider increasing the bitrate to 192k or 256k using the FFmpeg command.
WAV files can contain metadata in INFO chunks or ID3 tags embedded in the file, but support is inconsistent across software. WMA has its own metadata system using ASF (Advanced Systems Format) attributes. FFmpeg will attempt to map any recognized WAV metadata — such as title, artist, or album — to WMA tags during conversion, but unusual or non-standard WAV metadata fields may be dropped. You should verify tags in Windows Media Player or a tag editor like Mp3tag after conversion.
Replace the value after -b:a in the command to set a different bitrate. For example, to encode at 192k, use: ffmpeg -i input.wav -c:a wmav2 -b:a 192k output.wma. Supported practical options include 64k, 96k, 128k, 160k, 192k, 256k, and 320k. Higher bitrates preserve more audio detail but produce larger files; 128k is a reasonable default for most content, while 64k is suitable for voice-only audio where file size is critical.
Yes — wmav1 (Windows Media Audio version 1) can be selected with the flag -c:a wmav1 if you need compatibility with very old hardware or software that predates WMA v2. However, wmav2 has been the standard since the early 2000s and offers better audio quality at the same bitrate, so there is rarely a practical reason to choose wmav1 unless you are targeting a specific legacy device that explicitly requires it. The full command would be: ffmpeg -i input.wav -c:a wmav1 -b:a 128k output.wma.
A standard 16-bit stereo WAV at 44.1 kHz (CD quality) stores audio at approximately 1411 kbps. At the default 128k WMA bitrate, the output file will be roughly 10–11 times smaller. For example, a 50 MB WAV file would become approximately 4–5 MB in WMA. The exact ratio depends on the original WAV's bit depth, sample rate, and channel count — a 24-bit or 32-bit WAV will show even more dramatic size reduction.
Yes — on Linux or macOS, you can run: for f in *.wav; do ffmpeg -i "$f" -c:a wmav2 -b:a 128k "${f%.wav}.wma"; done. On Windows Command Prompt, use: for %f in (*.wav) do ffmpeg -i "%f" -c:a wmav2 -b:a 128k "%~nf.wma". This applies the same wmav2 encoding at 128k to every WAV file in the current directory and outputs a matching WMA file for each. The browser-based tool on this page processes one file at a time, so the FFmpeg command is the recommended approach for bulk conversions.
Technical Notes
The wmav2 codec used in this conversion is FFmpeg's implementation of Microsoft's Windows Media Audio v2 algorithm, which is a transform-based lossy codec operating in the frequency domain — broadly comparable to AAC or MP3 in its psychoacoustic approach, but not interchangeable with either. The output is wrapped in an ASF (Advanced Systems Format) container with the .wma extension, which is distinct from the WAV RIFF container. WMA supports DRM (Digital Rights Management) at the container level, but FFmpeg does not apply DRM during encoding — the output file will be DRM-free. One important limitation is that WMA is not universally supported outside the Windows ecosystem: macOS and iOS require QuickTime component additions or third-party apps to play WMA, and many web browsers cannot play WMA natively. If cross-platform compatibility is a concern, MP3 or AAC may be more suitable targets. Additionally, WAV files using non-PCM codecs (such as adpcm_ms or pcm_alaw) will still be correctly decoded by FFmpeg before re-encoding to WMA, as FFmpeg handles the input decoding transparently regardless of the WAV's internal codec.