Convert WMV to AMR — Free Online Tool
Convert WMV video files to AMR audio by extracting and re-encoding the speech-optimized audio track using the libopencore_amrnb codec. AMR's Adaptive Multi-Rate compression is purpose-built for voice content, making this conversion ideal when you need a compact, telephony-compatible audio file from a Windows Media Video source.
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FFmpeg Command
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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
During this conversion, the video stream from the WMV container is discarded entirely — AMR is a pure audio format with no video support. The audio track, typically encoded with WMA v2 (wmav2) inside the ASF container, is decoded from its Windows Media Audio compression and then re-encoded from scratch using the libopencore_amrnb encoder at the AMR-NB (Narrowband) standard. This is a full transcode, not a remux. AMR-NB operates at a fixed 8kHz sample rate and is mono-only, so any stereo WMA audio will be downmixed and resampled during encoding. The output .amr file contains only the speech-optimized audio stream at the selected bitrate, with the highest quality AMR-NB mode being 12,200 bits/sec (mode 7, also known as MR122).
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In this browser-based tool, the equivalent FFmpeg.wasm WebAssembly binary runs locally in your browser — no files leave your device. |
-i input.wmv
|
Specifies the input file: a WMV file using Microsoft's ASF container, typically containing an msmpeg4 video stream and a wmav2 audio stream. FFmpeg automatically demuxes the ASF container and identifies both streams. |
-c:a libopencore_amrnb
|
Sets the audio encoder to libopencore_amrnb, the open-source implementation of the AMR Narrowband codec. This re-encodes the decoded WMA audio from the WMV into AMR-NB format at 8kHz mono, which is the standard for GSM telephony and mobile voice applications. |
-b:a 12200
|
Sets the AMR-NB encoding mode to 12,200 bps, which corresponds to Mode 7 (MR122) — the highest quality mode in the AMR-NB standard. Unlike conventional codecs, AMR-NB bitrates are discrete codec modes, not freely variable values; 12200 is the ceiling for AMR-NB quality. |
output.amr
|
Defines the output file with the .amr extension, which tells FFmpeg to wrap the encoded libopencore_amrnb bitstream in the standard AMR file format. The video stream from the original WMV is implicitly dropped because the AMR format supports only audio. |
Common Use Cases
- Extracting a voice narration or spoken commentary from a WMV screen recording to create a mobile-compatible voice memo or voicemail file
- Converting recorded Windows Media Player lectures or presentations to AMR for playback on GSM mobile phones and older feature phones that natively support the format
- Preparing voice content from WMV interviews or dictation recordings for upload to telephony systems, IVR platforms, or VoIP applications that accept AMR-NB input
- Stripping the audio from a WMV video of a speech or announcement to create a compact AMR file for distribution over low-bandwidth mobile networks
- Converting Windows Media Video recordings of voice-only conference calls to AMR for archiving in a format aligned with 3GPP mobile communication standards
- Extracting a single spoken-word audio track from a WMV training video to feed into a speech recognition or transcription pipeline that requires AMR input
Frequently Asked Questions
Yes — this is a lossy-to-lossy transcode, so quality degradation is inevitable and compounded by two stages of compression. The WMA v2 audio in a WMV file is decoded first, and then re-encoded into AMR-NB, which is designed exclusively for speech at 8kHz narrowband quality. Music, sound effects, or any non-voice content will sound noticeably degraded or muffled in the output because AMR's compression model is optimized around the frequency range of human speech, not full-range audio.
AMR Narrowband (AMR-NB) is a telephony codec defined by the 3GPP standard for GSM voice calls, which are inherently mono channels. FFmpeg will automatically downmix a stereo WMA audio track from the WMV source to mono before encoding it into AMR-NB. You won't get an error, but any stereo spatial information or channel separation in the original WMV audio will be lost in the output.
AMR-NB uses a set of fixed codec modes rather than freely variable bitrates — 12,200 bps corresponds to Mode 7 (MR122), which is the highest quality mode available in the AMR-NB standard. The other valid bitrate values (4750, 5150, 5900, 6700, 7400, 7950, 10200) correspond to lower AMR-NB modes with increasingly aggressive speech compression. For voice recordings from WMV files, using 12200 preserves the most detail, though even at this setting the 8kHz sample rate cap means the output will sound narrowband compared to the original WMA audio.
Yes — FFmpeg supports AMR Wideband via the libopencore_amrwb codec, which operates at 16kHz instead of 8kHz and offers significantly better speech intelligibility and naturalness. To use it, you would change the codec flag in the command to '-c:a libopencore_amrwb' and select a valid AMR-WB bitrate. However, note that AMR-WB support is less universal than AMR-NB on older mobile devices, so consider your target playback environment before switching.
Replace the value after '-b:a' with any valid AMR-NB mode bitrate: 4750, 5150, 5900, 6700, 7400, 7950, 10200, or 12200. For example, to use the lowest-bitrate mode for maximum compression, the command becomes: ffmpeg -i input.wmv -c:a libopencore_amrnb -b:a 4750 output.amr. Keep in mind these are not arbitrary bitrates — FFmpeg will snap the value to the nearest valid AMR-NB mode, so only the eight listed values are meaningful for this codec.
The AMR file format has very limited metadata support compared to the ASF container used by WMV, which can carry rich Windows Media metadata fields. Most metadata from the WMV source — including title, author, description, and copyright tags — will not be carried over to the AMR output. The .amr format is a raw bitstream container focused purely on the encoded speech data, with no standardized metadata block equivalent to WMV's ASF Content Description Object.
Technical Notes
The WMV format uses Microsoft's Advanced Systems Format (ASF) container, and FFmpeg requires the '-f asf' flag when writing ASF/WMV output — though for reading WMV as input, FFmpeg detects the format automatically. The default audio codec inside WMV is WMA v2 (wmav2), which must be fully decoded before re-encoding to AMR-NB, as there is no direct transcoding shortcut between these two codec families. The libopencore_amrnb encoder used here is an open-source implementation of the 3GPP AMR-NB standard and must be compiled into your FFmpeg build — most standard FFmpeg distributions include it, but some minimal builds may not. AMR-NB strictly enforces mono audio at an 8kHz sample rate; if the WMV source contains audio at 44.1kHz or 48kHz stereo (common for WMV files with music or mixed content), FFmpeg will automatically resample and downmix before encoding, which is a significant quality-reducing transformation. The output .amr file size will be dramatically smaller than the original WMV — a 12,200 bps AMR-NB file is roughly 90KB per minute of audio, compared to a typical WMV which might include a 128kbps WMA audio track alongside video. This format is best suited exclusively for voice/speech content extracted from WMV sources.