Extract Audio from FLV to AAC — Free Online Tool
Extract the AAC audio track from an FLV file — the native audio codec used in Flash Video — and save it as a standalone .aac file. Because FLV files commonly encode audio in AAC already, this tool efficiently strips the video stream and repackages the audio with minimal quality impact, making it ideal for recovering audio from Flash-era web video.
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 (Flash Video) files typically contain either AAC or MP3 audio alongside H.264 or Sorenson Spark video. When extracting to AAC, FFmpeg discards the video stream entirely using the -vn flag, then re-encodes the audio using the native AAC encoder at the target bitrate. If the source audio was already AAC (the most common case in modern FLV files), the re-encoding step is technically a transcode within the same codec family rather than a copy, meaning there is a small generational quality cost — though at 128k this is rarely perceptible. The output is a raw .aac file (ADTS-framed AAC), not wrapped in an MP4 or M4A container, making it immediately usable in most audio players and streaming pipelines.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary. In the browser version of this tool, this runs via FFmpeg.wasm compiled to WebAssembly, so no files leave your device. |
-i input.flv
|
Specifies the input Flash Video file. FFmpeg reads both the video and audio streams from the FLV container, which may contain H.264 video alongside AAC or MP3 audio. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore the H.264 or Sorenson Spark video stream in the FLV and produce an audio-only output file. |
-c:a aac
|
Encodes the output audio using FFmpeg's built-in AAC encoder, producing AAC-LC format audio — the same codec family used by Apple devices, YouTube, and most modern streaming services. |
-b:a 128k
|
Sets the AAC audio bitrate to 128 kilobits per second, which is the standard default considered transparent for most audio content and produces files roughly 1 MB per minute of audio. |
output.aac
|
Writes the result as a raw ADTS-framed AAC bitstream file. The .aac extension tells FFmpeg to use the ADTS container format, which is directly playable by most media players and compatible with mobile devices and web browsers. |
Common Use Cases
- Recovering the audio commentary or narration from an old Flash-based tutorial or webinar recording saved as an FLV file
- Extracting a music track or DJ mix that was captured from a Flash-based streaming platform before it was shut down
- Pulling the audio from a Flash game trailer or promotional FLV to use in a video montage or presentation
- Converting FLV podcast episodes — common in early 2000s web video — into AAC audio files compatible with Apple Podcasts and modern mobile apps
- Archiving the audio portion of Flash video content before the files become completely unplayable as Flash support disappears from all platforms
- Extracting voiceover audio from FLV screen recordings to reuse in updated video content without re-recording
Frequently Asked Questions
Yes, there is a small quality loss because FFmpeg re-encodes the audio rather than stream-copying it directly. This happens because raw .aac (ADTS) output requires the audio to pass through the encoder pipeline. If you want truly lossless extraction from an AAC-encoded FLV, you could instead remux to M4A using '-c:a copy', though that output format is not what this tool targets. At 128k the quality difference from a single re-encode generation is extremely subtle and imperceptible in most listening contexts.
You can inspect your FLV file by running 'ffprobe input.flv' in a terminal, which will show the audio codec in the stream information — it will appear as 'aac' or 'mp3' (libmp3lame). FLV files produced by Adobe Flash Player 9 and later predominantly use AAC, while older FLV files often contain MP3 audio. If your source contains MP3 audio, the conversion to AAC will involve a cross-codec transcode, which carries slightly more quality risk than a same-codec transcode.
The output is a raw ADTS-framed AAC bitstream (.aac), which contains only the audio data with minimal container overhead. An .m4a file is AAC audio wrapped inside an MPEG-4 container, which adds support for metadata tags, chapter markers, and better seeking. For most playback purposes both work fine, but .m4a is generally preferred for music libraries and iTunes/Apple Music, while raw .aac is commonly used in streaming and broadcast pipelines. If you need .m4a specifically, the FFmpeg command can be adapted by changing the output filename to output.m4a.
Yes — AAC is Apple's preferred audio format and is fully compatible with all iOS and macOS devices, iTunes, and Apple Music. The raw .aac ADTS format produced by this tool plays natively on iPhones and iPads. If you intend to import the file into Apple Music or tag it with metadata, renaming or remuxing it to .m4a is recommended, as that container better supports ID3-style tags.
Replace the '-b:a 128k' value in the command with your desired bitrate. For example, use '-b:a 192k' for higher quality or '-b:a 96k' for a smaller file. AAC at 128k is considered transparent (indistinguishable from the original) for most listeners, but if your source FLV audio was encoded at a higher bitrate, extracting at 192k or 256k preserves more of the original quality ceiling. Note that increasing the bitrate beyond what the source was encoded at does not recover lost quality — it only increases file size.
Yes. On Linux or macOS, you can run a shell loop: 'for f in *.flv; do ffmpeg -i "$f" -vn -c:a aac -b:a 128k "${f%.flv}.aac"; done'. On Windows Command Prompt, use: 'for %f in (*.flv) do ffmpeg -i "%f" -vn -c:a aac -b:a 128k "%~nf.aac"'. This is especially useful when archiving a collection of FLV files from an old Flash-based video library, since the browser tool handles one file at a time.
Technical Notes
FLV is a legacy Adobe container format that was dominant in web video from roughly 2003 to 2015, before HTML5 video rendered Flash Player obsolete. The format supports a limited codec set: video in H.264 (libx264) or the older Sorenson Spark (flv codec), and audio in either AAC or MP3. When targeting AAC output, FFmpeg uses its built-in 'aac' encoder (not the higher-quality libfdk_aac, which requires a custom FFmpeg build). The built-in encoder produces compliant AAC-LC output that is broadly compatible across devices and platforms. One known limitation of raw .aac output is the absence of a seekable container, which can cause issues in some media players when scrubbing through long files — wrapping in M4A resolves this. Metadata embedded in the FLV (such as title or duration stored in the onMetaData tag) is not preserved in the raw .aac output, so any tagging should be applied post-conversion with a tool like FFmpeg's '-metadata' flag or a dedicated audio tagger. File sizes will be significantly smaller than the source FLV since the video stream — typically the largest component — is completely discarded.