Extract Audio from FLV to WMA — Free Online Tool

Extract audio from FLV Flash Video files and convert it to WMA format using the wmav2 codec — Microsoft's standard Windows Media Audio codec. This tool strips the video stream entirely and re-encodes the audio track to WMA, making your Flash Video audio compatible with Windows Media Player, Xbox, and legacy Microsoft ecosystem software.

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Estimated output:

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How It Works

FLV files typically carry audio encoded in either AAC or MP3 (via libmp3lame). Since WMA uses Microsoft's proprietary wmav2 codec, a direct stream copy is not possible — the audio must be fully decoded from its original FLV codec and re-encoded into wmav2. The video stream is discarded entirely using the -vn flag, so none of the visual data is processed or included in the output. The result is a standalone .wma audio file at 128k bitrate by default, encoded with wmav2, which is broadly compatible with Windows Media Player and Windows-native applications. Because both the input and output formats are lossy, this is a lossy-to-lossy transcode, meaning some audio generation loss is introduced relative to the original FLV audio.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg program, the open-source multimedia processing engine that handles all decoding, filtering, and encoding in this conversion. This tool runs FFmpeg compiled to WebAssembly (FFmpeg.wasm) directly in your browser.
-i input.flv Specifies the input file — in this case an FLV Flash Video file. FFmpeg reads both the video stream and the audio stream (AAC or MP3) from this container, though only the audio will be used for the WMA output.
-vn Stands for 'video none' — this flag tells FFmpeg to completely ignore and discard the video stream from the FLV file. Since the output is a WMA audio-only file, no video data is processed or written, which speeds up the conversion significantly.
-c:a wmav2 Sets the audio codec to wmav2, Microsoft's second-generation Windows Media Audio codec. This re-encodes the FLV's AAC or MP3 audio into the wmav2 format required by the WMA container — a direct stream copy is not possible here because the source and destination codecs are fundamentally different.
-b:a 128k Sets the audio bitrate to 128 kilobits per second for the wmav2 encoder. This is the default balance between file size and audio quality for WMA; increase to 192k or 256k for better fidelity on music content, or reduce to 96k for smaller files where audio quality is less critical.
output.wma Specifies the output filename and format. The .wma extension tells FFmpeg to use the ASF (Advanced Systems Format) container that WMA files require, and the wmav2-encoded audio stream is written into this file as the final product.

Common Use Cases

  • Extracting commentary or narration from a Flash-era tutorial video to archive or repurpose it as a standalone WMA audio file for Windows Media Player playlists
  • Converting FLV recordings from legacy Flash-based webinar or screen capture tools into WMA for distribution via older Windows-centric intranets or corporate media servers that require WMA format
  • Pulling the audio track from an FLV music video downloaded from an early-2000s video site to create a WMA file compatible with a Windows Media-based portable media player or Zune device
  • Stripping audio from FLV game recordings or livestream archives to produce WMA commentary files for use in Windows Movie Maker or legacy video editing projects
  • Archiving audio from Flash-based educational content before the platform becomes entirely inaccessible, outputting to WMA for integration into a Windows-based digital library system
  • Extracting background music or soundtracks embedded in FLV promotional videos to produce WMA files for Windows-native media kiosks or digital signage systems

Frequently Asked Questions

Yes, some quality loss is unavoidable. FLV typically stores audio as AAC or MP3, both of which are lossy formats. WMA (wmav2) is also lossy, so this conversion is a lossy-to-lossy transcode — the audio is decoded from its original FLV codec and re-encoded using wmav2, introducing an additional generation of compression artifacts. To minimize this loss, use the highest bitrate option available (320k). If you have access to an uncompressed or lossless source, convert from that instead of from FLV.
WMA files require audio encoded specifically with Microsoft's wmav2 (or wmav1) codec. FLV files carry audio in AAC or MP3 format, neither of which is compatible with the WMA container. Because the codec formats are fundamentally incompatible, FFmpeg must fully decode the source audio and re-encode it as wmav2 — there is no stream-copy shortcut for this conversion.
Modify the -b:a flag value in the command. For example, replace -b:a 128k with -b:a 256k for higher quality, or -b:a 96k for a smaller file size. WMA with wmav2 supports bitrates from 64k up to 320k. Keep in mind that going below 96k with wmav2 can produce noticeably degraded audio, while 192k or higher is generally considered good quality for music content.
It depends on what metadata was present in the FLV file and whether FFmpeg can map it to WMA's tag structure. FLV has limited native metadata support, so most FLV files carry minimal or no embedded tags. WMA supports rich metadata through its ASF container (title, artist, album, genre, etc.), but if the source FLV lacks these fields they will not appear in the output. You can add metadata manually by appending flags like -metadata title='My Title' to the FFmpeg command.
wmav2 is the better choice for virtually all use cases and is what this tool uses by default. wmav2 produces noticeably better audio quality than wmav1 at the same bitrate, and it has been the standard WMA codec since Windows Media Player 7. wmav1 is an older, lower-quality codec that should only be used if you need compatibility with extremely outdated hardware or software that specifically cannot decode wmav2.
The single command shown converts one file at a time, but you can adapt it for batch processing in a terminal. On Linux or macOS, use: for f in *.flv; do ffmpeg -i "$f" -vn -c:a wmav2 -b:a 128k "${f%.flv}.wma"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a wmav2 -b:a 128k "%~nf.wma". This processes every FLV in the current directory and outputs a corresponding WMA file for each.

Technical Notes

FLV (Flash Video) is an Adobe container format that encodes audio as either AAC or MP3 using libmp3lame, both lossy codecs. WMA uses Microsoft's proprietary ASF (Advanced Systems Format) container with wmav2 audio encoding, which is also lossy but uses a completely different psychoacoustic model than AAC or MP3. Because the codecs are incompatible, transcoding is mandatory and introduces generation loss — this is an inherent limitation of converting between two lossy formats. The wmav2 codec performs well at mid-to-high bitrates (128k–320k) but can struggle with transient audio at lower bitrates compared to modern AAC encoders. WMA files support DRM (Digital Rights Management) at the container level, though FFmpeg does not apply DRM during encoding, so the output will be an unprotected WMA file. FLV does not support subtitles, chapters, or multiple audio tracks, so none of these concerns apply to this conversion. One known limitation: very old FLV files from early Flash Player versions may use Sorenson Spark video with raw PCM or ADPCM audio — in those cases FFmpeg will still decode the audio correctly before re-encoding to wmav2, but quality from such low-fidelity sources will be limited by the original recording.

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