Extract Audio from WMV to AMR — Free Online Tool
Extract audio from WMV video files and convert it to AMR format, re-encoding the WMV's wmav2 audio stream through the libopencore_amrnb codec — a speech-optimized encoder originally designed for mobile telephony. This is especially useful for pulling voice content, narrations, or meeting recordings from Windows Media Video files into a format compatible with mobile and telephony systems.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your WMV 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
WMV files store audio using Microsoft's wmav2 codec (or occasionally AAC or MP3) inside an Advanced Systems Format (ASF) container. AMR has no compatible overlap with any WMV audio codec, so a full transcode is required — the wmav2 audio stream is decoded to raw PCM and then re-encoded using the libopencore_amrnb encoder into Adaptive Multi-Rate Narrowband format. The video stream is discarded entirely using the -vn flag, since AMR is an audio-only container. AMR's encoder is tuned specifically for speech frequencies and operates at very low bitrates (the default here is 12200 bps, the highest AMR-NB mode), so music or complex audio from WMV files will lose significant fidelity. The output is a mono, 8kHz-sampled AMR file regardless of the original WMV audio's channel count or sample rate.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg media processing engine. In the browser, this runs via FFmpeg.wasm compiled to WebAssembly — the same command works identically in a local FFmpeg installation for files over 1GB. |
-i input.wmv
|
Specifies the input WMV file. FFmpeg reads it as an ASF (Advanced Systems Format) container and identifies the enclosed wmav2 audio and msmpeg4 video streams for processing. |
-vn
|
Disables video output entirely, discarding the msmpeg4 video stream from the WMV. Since AMR is an audio-only format, this flag is required to prevent FFmpeg from attempting to encode video into a container that cannot hold it. |
-c:a libopencore_amrnb
|
Selects the libopencore_amrnb encoder to transcode the WMV's wmav2 audio into AMR Narrowband format. This encoder resamples the audio to 8kHz mono and applies the AMR-NB speech compression algorithm, which is optimized for voice intelligibility at very low bitrates. |
-b:a 12200
|
Sets the AMR-NB encoding mode to 12200 bps, the highest-quality mode available in the AMR-NB standard (corresponding to AMR mode 7). This produces the most intelligible speech output from the WMV source audio while remaining within AMR's narrowband constraints. |
output.amr
|
Defines the output filename with the .amr extension, which tells FFmpeg to write the encoded audio into a raw AMR file container. The AMR container is bare-bones — it holds only the encoded frames with a simple file header and no metadata fields. |
Common Use Cases
- Extracting voice narrations from WMV screen recordings or training videos for use in IVR (interactive voice response) telephony systems that require AMR input
- Pulling meeting or conference call audio captured as WMV files and converting it to AMR for storage or playback on older mobile handsets
- Converting Windows Media Player–compatible video voicemails or recorded announcements into AMR for upload to mobile network infrastructure
- Archiving the speech content of legacy WMV corporate presentations into a compact AMR format optimized for low-bandwidth voice transmission
- Extracting audio from WMV police or security camera recordings for submission to telephony-based transcription or voice analysis pipelines that consume AMR files
Frequently Asked Questions
Only speech content will translate reasonably well. AMR-NB encodes audio at 8kHz mono, which is sufficient for voice intelligibility but discards most of the frequency range present in WMV's wmav2 audio (which typically supports up to 22kHz). Music, sound effects, or any stereo content in the WMV will be downmixed to mono and heavily degraded. If the WMV contains a narration or spoken-word track, the result will be clear enough for telephony use.
AMR-NB (libopencore_amrnb) is inherently a mono codec — it was designed for single-channel mobile voice transmission and does not support stereo encoding. FFmpeg automatically downmixes the WMV's audio channels to mono during the transcode. If your WMV has a true stereo mix with distinct left/right content, that spatial information will be lost in the AMR output.
These numbers are bitrates in bits per second corresponding to the AMR-NB codec's fixed encoding modes, ranging from 4750 bps (most compressed, lowest quality) to 12200 bps (least compressed, highest quality). For speech extracted from a WMV file, 12200 bps (the default used here) gives the most intelligible result. Dropping to 4750 or 5150 is only advisable if file size is critical and the speech is very clear, as lower modes introduce noticeable artifacts in voice recordings.
WMV files can carry metadata in the ASF container (title, author, copyright, etc.), but the AMR format has extremely limited metadata support. The libopencore_amrnb encoder does not write standard tag fields into the output AMR file, so metadata from the source WMV will not be present in the converted file. If metadata preservation is important, consider converting to a format like M4A or MP3 instead.
Replace the value after -b:a in the command with one of the valid AMR-NB mode bitrates: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200. For example, to use the lowest bitrate mode, run: ffmpeg -i input.wmv -vn -c:a libopencore_amrnb -b:a 4750 output.amr. Note that AMR-NB does not support arbitrary bitrates — FFmpeg will snap the value to the nearest valid mode, so using values outside this list may produce unexpected results.
Yes. On Linux or macOS you can loop over files with: for f in *.wmv; do ffmpeg -i "$f" -vn -c:a libopencore_amrnb -b:a 12200 "${f%.wmv}.amr"; done. On Windows Command Prompt, use: for %f in (*.wmv) do ffmpeg -i "%f" -vn -c:a libopencore_amrnb -b:a 12200 "%~nf.amr". This browser-based tool processes one file at a time, so the FFmpeg command is particularly useful for handling large batches locally.
Technical Notes
The WMV-to-AMR conversion involves two fundamentally different audio technologies with almost no common ground. WMV's default wmav2 codec is a wideband, perceptual audio codec designed for general-purpose media (music, voice, effects) at bitrates from 64k to 320k bps and sample rates up to 48kHz stereo. AMR-NB via libopencore_amrnb is a narrowband, speech-optimized codec fixed at 8kHz mono — specifications inherited from GSM cellular standards. FFmpeg must resample the WMV audio from its native sample rate down to 8000 Hz and downmix channels to mono before encoding, which are lossy operations applied on top of the AMR encode itself. The output file size will be dramatically smaller than the source WMV — a 128k wmav2 stream at 12200 bps AMR-NB represents roughly a 10x reduction in audio bitrate. WMV files that use DRM (Digital Rights Management) cannot be processed by FFmpeg or this tool, as the encrypted content cannot be decoded without the license key. Additionally, if a WMV file contains multiple audio tracks (a supported feature of the ASF container), FFmpeg will select the first audio track by default; use -map 0:a:1 to select an alternate track before running the conversion.