Convert WMA to WAV — Free Online Tool
Convert WMA files to WAV by decoding Microsoft's proprietary WMV2-encoded audio into uncompressed PCM (16-bit signed little-endian) — producing a lossless-quality WAV file with maximum compatibility across any audio software, DAW, or broadcast workflow.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
WMA files store audio using Microsoft's proprietary WMA codec (typically wmav2), which is a lossy compression format similar in concept to MP3 or AAC. During conversion, FFmpeg fully decodes the compressed wmav2 bitstream back into raw audio samples, then writes those samples into a WAV container using the pcm_s16le codec — 16-bit signed PCM in little-endian byte order. This is a full decode-and-re-encode operation, not a remux: the lossy compression is removed entirely and replaced with uncompressed audio. The resulting WAV file is bit-for-bit identical in PCM values to what the WMA decoder produces, meaning no additional quality is lost beyond what was already lost when the original WMA was encoded. WAV imposes no further compression, so the output file will be significantly larger than the source WMA.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that handles decoding the WMA input and encoding the WAV output in this conversion. |
-i input.wma
|
Specifies the input file — a WMA audio file typically encoded with the wmav2 codec. FFmpeg automatically detects the WMA container and selects the appropriate decoder without requiring you to specify the input format manually. |
-c:a pcm_s16le
|
Sets the audio codec for the output to pcm_s16le — 16-bit signed PCM in little-endian byte order. This decodes the lossy wmav2 audio entirely and writes raw, uncompressed audio samples into the WAV container, which is the standard format expected by virtually all audio software and hardware. |
output.wav
|
Defines the output filename with a .wav extension, which tells FFmpeg to wrap the pcm_s16le audio stream in a WAV container — Microsoft's standard uncompressed audio file format originally developed alongside IBM. |
Common Use Cases
- Importing WMA recordings from Windows Media Player or older Windows devices into a DAW like Pro Tools, Logic Pro, or Ableton, which often require uncompressed WAV for reliable editing and time-stretching.
- Preparing audio assets for broadcast or post-production pipelines that mandate uncompressed PCM WAV as the delivery format, rejecting proprietary lossy codecs like WMA.
- Unlocking DRM-free WMA files for archival — once decoded to WAV, the audio is stored in an open, universally readable format that won't break if Microsoft discontinues WMA support.
- Converting WMA voicemail recordings or dictation files exported from Windows systems into WAV so they can be transcribed using speech-to-text tools like Whisper or cloud APIs that prefer PCM WAV input.
- Stripping WMA compression from audio samples or music beds before mastering, ensuring no codec artifacts compound when the file is re-processed or re-exported.
- Making WMA-format audio files compatible with hardware devices, video editors, or legacy software that reads WAV but has no WMA decoder.
Frequently Asked Questions
No — converting from WMA to WAV does not recover quality that was lost when the WMA was originally encoded. WMA is a lossy codec, meaning audio data was permanently discarded during compression. The WAV output will be an exact, uncompressed representation of what the WMA decoder produces, but any artifacts introduced by the original wmav2 encoding — such as smearing of high frequencies or subtle distortion at low bitrates — are baked in and cannot be reversed. The conversion is lossless from the WMA onward, but the WMA itself was never lossless.
WMA uses lossy compression to reduce file size — a typical 128 kbps WMA is roughly 10–15 times smaller than an equivalent uncompressed WAV. When the WMA is decoded to pcm_s16le WAV, the audio is stored as raw 16-bit samples at the full sample rate (commonly 44100 Hz stereo), which requires approximately 10 MB per minute. This is expected and correct behavior: WAV stores every sample explicitly with no compression, making it much larger but also instantly readable by any audio software without a proprietary decoder.
Partially. FFmpeg will attempt to copy metadata tags from the WMA container into the WAV file's INFO chunk during conversion, but WAV has very limited native metadata support compared to WMA. Standard fields like title and artist may transfer, but WMA-specific or extended tags may be dropped. If metadata preservation is critical, consider using a dedicated tag editor after conversion, or evaluate whether a format like FLAC (which also supports lossless encoding) would better suit your tagging needs.
The flag '-c:a pcm_s16le' tells FFmpeg to encode the audio using 16-bit signed PCM in little-endian byte order, which is the standard WAV format used by virtually all consumer and professional software. If your source WMA was encoded from high-bit-depth audio and you need more precision, you could substitute 'pcm_s24le' for 24-bit PCM or 'pcm_s32le' for 32-bit — for example: ffmpeg -i input.wma -c:a pcm_s24le output.wav. For most WMA files at 128–192 kbps, pcm_s16le is more than sufficient since the lossy source doesn't contain meaningful audio data beyond 16 bits of dynamic range.
Yes. On Linux or macOS, you can loop over all WMA files in a directory with: for f in *.wma; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.wma}.wav"; done. On Windows Command Prompt, use: for %f in (*.wma) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". Each WMA file will be decoded and written to a corresponding WAV file with the same base name. This is especially practical for large collections that exceed the 1 GB browser limit of this online tool.
If your goal is maximum compatibility — especially with hardware devices, video editing software, or broadcast systems — WAV is the safer choice since it is universally supported without any decoder. If you want to reduce file size while still keeping lossless PCM audio, FLAC is a better archival format: it compresses the same PCM data losslessly to roughly 50–60% of the WAV size and supports rich metadata. The FFmpeg command for FLAC would be: ffmpeg -i input.wma -c:a flac output.flac. Either way, no quality beyond the original WMA lossy encoding is recovered.
Technical Notes
The default WMA audio codec is wmav2, a proprietary Microsoft codec based on a modified discrete cosine transform (MDCT) similar to AAC and Vorbis, though generally considered slightly inferior in quality at equivalent bitrates. FFmpeg's wmav2 decoder has excellent compatibility with standard WMA files, but WMA files protected with Microsoft's DRM (PlayReady or the older Windows Media DRM) cannot be decoded by FFmpeg and will fail to convert — DRM removal requires licensed Microsoft software or a DRM-free source file. The output codec pcm_s16le is the canonical WAV format: 16-bit signed integers, little-endian byte order, stored at whatever sample rate and channel count the WMA source uses (commonly 44100 Hz stereo). No sample rate conversion or channel remixing is applied by this command unless explicitly added. WAV files do not support chapters, embedded subtitles, or multiple audio tracks, and neither does WMA, so there is no feature loss in that regard. One known limitation: WAV files are capped at approximately 4 GB by the standard 32-bit chunk size header, though FFmpeg can write RF64-extended WAV for longer files if needed.