Convert AMR to WMA — Free Online Tool
Convert AMR audio files — the compressed speech format used in mobile voice recordings and telephony — to WMA format using the wmav2 codec, making your recordings compatible with Windows Media Player and Microsoft-ecosystem applications. This tool runs entirely in your browser via FFmpeg.wasm, so your voice memos and call recordings never leave your device.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your AMR file here
or click to browse
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
AMR (Adaptive Multi-Rate) audio is specifically engineered for speech compression at very low bitrates — as low as 4.75 kbps — using codecs like libopencore_amrnb (narrowband) optimized for the 300–3400 Hz frequency range of human voice. Converting to WMA requires a full audio decode-and-reencode: the AMR stream is decoded from its speech-optimized encoding into raw PCM audio, then re-encoded using the wmav2 codec at a much higher bitrate (128k by default). Because AMR narrowband captures only the speech frequency range at a low sample rate (typically 8 kHz), the WMA output will not gain quality beyond what the original AMR captured — the wider frequency range WMA supports simply won't be filled with meaningful data. The result is a significantly larger file that trades AMR's extreme compression for broad compatibility with Windows software and media players.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing engine. In this browser-based tool, FFmpeg runs as a WebAssembly (wasm) binary entirely within your browser — no server processes this command. |
-i input.amr
|
Specifies the input AMR file. FFmpeg detects the AMR container and selects the appropriate decoder — libopencore_amrnb for narrowband AMR or libopencore_amrwb for wideband AMR-WB — to decode the speech-compressed audio into raw PCM for re-encoding. |
-c:a wmav2
|
Sets the audio encoder to wmav2, Microsoft's Windows Media Audio version 2 codec. wmav2 is the standard choice for WMA output, offering better quality than wmav1 at equivalent bitrates and compatibility with all modern Windows Media Player versions and WMA-capable devices. |
-b:a 128k
|
Sets the audio bitrate for the WMA output to 128 kbps. This is roughly 10 times the maximum bitrate of the AMR source (12.2 kbps), which ensures the WMA encoder has sufficient headroom to represent the decoded speech audio — though the perceptible quality ceiling remains limited by the original AMR recording's narrow frequency range. |
output.wma
|
Specifies the output filename with the .wma extension, which signals FFmpeg to wrap the wmav2-encoded audio in the ASF (Advanced Systems Format) container — the standard container used by all WMA files and required for compatibility with Windows Media Player and WMA-aware software. |
Common Use Cases
- Converting voice memos recorded on older Nokia or Android phones (saved as .amr) into WMA so they can be organized and played in Windows Media Player or imported into Windows-based audio editors
- Preparing mobile call recordings in AMR format for use in corporate Windows environments where WMA is the standard audio format for documentation and archiving
- Transcribing AMR voice notes by first converting them to WMA for compatibility with Windows-native speech-to-text or transcription software that doesn't accept AMR input
- Packaging AMR field recordings or interviews into WMA for distribution on Windows-centric intranet portals or older digital audio players that support WMA but not AMR
- Converting AMR audio attachments received via MMS or messaging apps into a WMA format compatible with Windows desktop editing tools for cleanup and noise reduction
Frequently Asked Questions
No — converting AMR to WMA will not recover audio quality that wasn't captured in the original file. AMR narrowband encodes only the 300–3400 Hz speech frequency range at 8 kHz sample rate, and that limitation carries through to the WMA output. The WMA file will be much larger and more compatible, but the audio fidelity ceiling is set by the original AMR source. If your AMR recording sounds thin or telephone-quality, the WMA version will sound the same.
AMR is an extreme speech-compression format designed for mobile networks, operating at bitrates as low as 4.75 kbps to 12.2 kbps. The default WMA output uses a 128 kbps bitrate — roughly 10 to 27 times higher than a typical AMR file. This dramatic size increase is entirely expected and reflects WMA's much higher bitrate, not added audio content. If file size is a concern, you can lower the WMA bitrate to 64k in the FFmpeg command without any perceptible quality difference given the limited fidelity of the AMR source.
AMR files have very limited or no standardized metadata tag support — most AMR recordings from phones contain no embedded tags at all. WMA, by contrast, uses the ASF container which supports rich metadata including title, artist, album, and comment fields. Since the AMR source almost certainly has no metadata, the WMA output will also have empty tags. You can add metadata separately after conversion using a tag editor or by modifying the FFmpeg command with flags like -metadata title='My Recording'.
Replace the -b:a 128k value with any supported WMA bitrate: 64k, 96k, 128k, 160k, 192k, 256k, or 320k. For example, use ffmpeg -i input.amr -c:a wmav2 -b:a 64k output.wma for a smaller file. Given that AMR narrowband source material tops out at 12.2 kbps and covers only the speech frequency range, there is little practical benefit to using bitrates above 96k or 128k — the extra data won't represent additional audio information from the source.
Yes — if you need compatibility with very old Windows Media Player versions (pre-WMP 9) or legacy hardware devices, you can substitute -c:a wmav1 in the FFmpeg command. However, wmav2 is supported by virtually all WMP versions since Windows XP and is the recommended choice. wmav1 also tends to produce lower quality output at the same bitrate compared to wmav2, so there is no quality reason to prefer it on modern systems.
The single-file command shown handles one file at a time, but you can adapt it for batch processing on your desktop. On Windows, use a for loop in Command Prompt: for %f in (*.amr) do ffmpeg -i "%f" -c:a wmav2 -b:a 128k "%~nf.wma". On macOS or Linux, use: for f in *.amr; do ffmpeg -i "$f" -c:a wmav2 -b:a 128k "${f%.amr}.wma"; done. The browser-based tool processes one file at a time, but there is no file size limit for batch scripts run locally.
Technical Notes
AMR narrowband (libopencore_amrnb) operates at a fixed 8 kHz sample rate and mono channel output, targeting the human speech frequency band. When re-encoding to WMA via wmav2, FFmpeg will upsample the audio — but upsampling does not synthesize frequencies that were never recorded. The resulting WMA file will technically have a higher sample rate container but the spectral content above ~3.4 kHz will be essentially empty or contain only interpolation artifacts. WMA via the ASF container does not support multiple audio tracks or chapter markers, which is consistent with AMR's own limitations. One notable consideration: AMR wideband (libopencore_amrwb), used in HD Voice calls, records at 16 kHz and covers up to 7 kHz — if your source is an AMR-WB file, the WMA output will carry meaningfully more audio information. The wmav2 codec is a lossy encoder and this conversion is lossy-to-lossy, meaning each transcode generation degrades audio slightly; avoid repeated re-encoding cycles. DRM features available in the WMA/ASF container are not applied by this tool or the FFmpeg command shown.