Extract Audio from FLV to OGG — Free Online Tool
Extract audio from FLV video files and save it as OGG using the open Vorbis codec — ideal for converting legacy Flash Video content into a patent-free, broadly supported audio format. The audio stream is re-encoded from AAC or MP3 (as found in most FLV files) into Vorbis at a balanced quality setting, discarding the video entirely.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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
FLV files typically contain video encoded with H.264 or the older FLV codec, paired with audio in AAC or MP3 format. This tool strips the video stream entirely and re-encodes the audio into the Ogg Vorbis format — a free, open-source lossy codec developed by Xiph.Org. Because FLV's native audio codecs (AAC, MP3) are not natively stored in OGG containers, a full audio transcode is required rather than a simple stream copy. The Vorbis encoder uses a variable bitrate quality scale (0–10), with the default quality level of 4 targeting roughly 128–160 kbps — comparable in perceptual quality to the 128k AAC or MP3 audio commonly found in FLV files. The result is a standalone .ogg audio file with no video data.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles all multimedia decoding, processing, and encoding. In this browser-based tool, FFmpeg runs locally via WebAssembly (ffmpeg.wasm) — your FLV file never leaves your device. |
-i input.flv
|
Specifies the input file — your FLV (Flash Video) source. FFmpeg will parse the FLV container to identify and demux the video and audio streams inside before processing begins. |
-vn
|
Stands for 'video none' — tells FFmpeg to discard the video stream entirely and not include it in the output. Since the goal is audio extraction from the FLV, the H.264 or FLV video track is dropped here. |
-c:a libvorbis
|
Selects the Vorbis encoder (libvorbis) to re-encode the audio into the OGG-compatible Vorbis format. Because FLV audio (AAC or MP3) cannot be stored natively in an OGG container, a full re-encode through libvorbis is required. |
-q:a 4
|
Sets the Vorbis variable bitrate quality level to 4 on a 0–10 scale, targeting approximately 128–160 kbps. This is a good default that balances file size and audio fidelity, roughly matching the quality of the 128k AAC or MP3 audio typically found in FLV files. |
output.ogg
|
Defines the output filename and tells FFmpeg to write the result as an OGG container. The .ogg extension signals FFmpeg to use the OGG muxer, which wraps the Vorbis audio stream in Xiph.Org's open container format. |
Common Use Cases
- Recover the audio commentary or narration from an old Flash-based tutorial or e-learning video saved as FLV
- Extract music or a DJ set recorded as a Flash livestream archive into an OGG file for playback in open-source media players like VLC or Audacious
- Convert FLV video podcasts from early web platforms (like pre-YouTube Flash embeds) into OGG audio for archiving in a patent-free format
- Strip the audio from a Flash game cutscene or promotional FLV file to use as a standalone sound asset in an open-source project
- Extract background music from a legacy FLV screensaver or web animation for use in a Linux-based multimedia workflow that prefers royalty-free formats
- Prepare audio content from FLV broadcast clips for upload to platforms like Wikipedia/Wikimedia Commons that require open-format media
Frequently Asked Questions
Yes, some quality loss occurs because this conversion involves transcoding between two lossy codecs — the source audio in FLV (typically AAC or MP3) is decoded and then re-encoded as Vorbis. This is sometimes called a 'generation loss' transcode. At the default quality level of 4 (roughly 128–160 kbps variable bitrate), the result is perceptually close to the original for most listeners, but if the source FLV audio was already heavily compressed at a low bitrate, artifacts may become more noticeable after re-encoding.
The output uses the Vorbis codec inside an OGG container — specifically encoded by FFmpeg's libvorbis library. OGG is just the container format; it can also hold Opus or FLAC audio, but this tool defaults to Vorbis because it offers the best balance of compatibility and quality for general use. Most modern media players including VLC, Firefox, Chrome, and Android's default player can decode OGG Vorbis natively.
FLV files support limited metadata, usually embedded as AMF (Action Message Format) tags containing fields like title, author, or duration. FFmpeg will attempt to map recognized metadata fields to OGG's Vorbis comment tags during conversion, but FLV metadata is often sparse or nonstandard, so fields may not transfer completely. OGG/Vorbis supports rich metadata including title, artist, album, and tracknumber tags, so you can manually add or edit these after conversion using a tag editor like MusicBrainz Picard or EasyTag.
Adjust the -q:a value to control Vorbis encoding quality. The scale runs from 0 (lowest, roughly 64 kbps) to 10 (highest, roughly 500 kbps), with 4 being the default and a good general-purpose setting. For example, use '-q:a 6' for higher quality at around 192–256 kbps, or '-q:a 2' to produce a smaller file at around 80–96 kbps. Unlike fixed bitrate encoding, Vorbis's -q:a uses variable bitrate, so the final file size will vary depending on the complexity of the audio content.
The displayed command processes one file at a time, but you can adapt it for batch processing in a terminal. On Linux or macOS, use a shell loop: 'for f in *.flv; do ffmpeg -i "$f" -vn -c:a libvorbis -q:a 4 "${f%.flv}.ogg"; done'. On Windows Command Prompt, use: 'for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a libvorbis -q:a 4 "%~nf.ogg"'. This is especially useful for archiving a folder full of legacy FLV recordings in one pass.
OGG Vorbis is a fully open and patent-free format, making it the preferred choice for projects that avoid proprietary codecs — including open-source software, Wikipedia media uploads, and Linux-centric workflows. At equivalent perceptual quality, Vorbis generally produces smaller files than MP3 and is competitive with AAC. The trade-off is slightly narrower hardware support: dedicated audio devices and some older phones may not play OGG natively, whereas MP3 is universally supported. If broad hardware compatibility matters more than open licensing, extracting to MP3 instead may be preferable.
Technical Notes
FLV was designed as a streaming container for Adobe Flash Player and typically encodes audio as either AAC (in F4V-compatible FLV files) or MP3 (in older FLV files). Neither of these codecs can be placed directly into an OGG container, so FFmpeg must fully decode the source audio and re-encode it as Vorbis — there is no lossless remux path for this conversion. The Vorbis encoder in FFmpeg (libvorbis) uses a quality-based variable bitrate mode by default, which means the output bitrate adapts to the complexity of the audio signal rather than targeting a fixed number. OGG supports chapter markers and multiple audio tracks in its specification, though these features are not applicable when extracting a single audio stream from a standard FLV source. One known limitation is that FLV files with video-only streams or corrupted audio tracks may produce an empty or silent OGG file — FFmpeg will typically warn about this in its console output. Additionally, FLV files that use on2VP6 video with Nellymoser or ADPCM audio (less common, found in very old Flash content) may require additional FFmpeg build flags to decode correctly, which the WebAssembly build used here fully supports.