Extract Audio from FLV to CAF — Free Online Tool
Extract audio from FLV (Flash Video) files and save it as CAF (Core Audio Format) — Apple's professional-grade container designed for high-resolution audio. The extracted audio is decoded from AAC or MP3 and re-encoded to uncompressed PCM (16-bit), making it immediately ready for editing in Logic Pro, GarageBand, or any macOS audio tool.
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 carry either AAC or MP3 audio alongside H.264 or legacy FLV video. During this conversion, FFmpeg discards the video stream entirely using the -vn flag, then decodes the compressed audio from the FLV container and re-encodes it as 16-bit signed little-endian PCM (pcm_s16le) wrapped in a CAF container. CAF was developed by Apple specifically to overcome the 4GB file size ceiling of WAV and AIFF, and it supports the same uncompressed PCM format used in professional Apple audio workflows. Because the audio is transcoded from lossy (AAC or MP3) to uncompressed PCM, no additional quality is gained — but the result is a lossless representation of the audio as it existed in the FLV, with no further lossy encoding artifacts introduced.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser-based version of this tool, FFmpeg runs as a WebAssembly (WASM) binary entirely within your browser — no files leave your machine. |
-i input.flv
|
Specifies the input FLV file. FFmpeg reads the FLV container, which typically carries an H.264 or legacy FLV video stream alongside an AAC or MP3 audio stream. |
-vn
|
Disables video output entirely — 'vn' stands for 'video none'. This drops the H.264 or FLV video track from the FLV source, ensuring only the audio stream is processed and written to the CAF file. |
-c:a pcm_s16le
|
Sets the audio codec to 16-bit signed little-endian PCM — uncompressed audio in the format natively expected by Apple's Core Audio framework. The AAC or MP3 audio from the FLV is fully decoded and stored as raw PCM samples, making the output immediately editable in Logic Pro or GarageBand without any further decoding step. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kilobits per second. For the pcm_s16le codec used here this parameter has no functional effect — uncompressed PCM bitrate is fixed by sample rate and bit depth — but it is included for consistency with the tool's quality settings interface. |
output.caf
|
Defines the output file as a CAF (Core Audio Format) container. FFmpeg infers the CAF format from the .caf extension and wraps the decoded PCM audio in this Apple-native container, which supports large file sizes and is compatible with macOS system audio APIs. |
Common Use Cases
- Extracting audio from Flash-era video lectures or screencasts originally published as FLV to use in Logic Pro or GarageBand projects
- Recovering narration tracks from archived FLV marketing or training videos for re-editing in Apple's pro audio tools
- Stripping the audio from an FLV livestream recording to create a clean podcast episode master file in CAF for further processing
- Converting FLV audio tracks to CAF for import into Core Audio-based iOS or macOS development projects requiring uncompressed audio assets
- Archiving audio content from legacy FLV files into a non-proprietary, Apple-native lossless container before the FLV source becomes unplayable
- Preparing extracted FLV commentary or voiceover tracks as PCM CAF files for alignment and mixing in Final Cut Pro audio workflows
Frequently Asked Questions
The FLV file's audio (typically AAC at 128k or MP3) is already lossy-compressed, and that compression is permanent — converting to uncompressed PCM in CAF does not recover detail that was discarded during the original encoding. What this conversion does guarantee is that no additional lossy encoding step is applied: the audio is decoded once and stored as raw PCM, so what you get in the CAF file is a faithful, lossless representation of the audio quality that existed in the FLV.
CAF does support AAC, so a stream-copy is technically possible, but the tool defaults to pcm_s16le because it produces the most universally editable output for Apple workflows — Logic Pro, GarageBand, and Core Audio APIs all work natively with uncompressed PCM. Additionally, the AAC stream inside an FLV is packaged differently than the AAC expected in a CAF container, so a raw copy would often require re-muxing anyway. Decoding to PCM sidesteps any compatibility issues entirely.
To use AAC instead of PCM, replace '-c:a pcm_s16le' with '-c:a aac' and keep '-b:a 128k' to control bitrate. For lossless FLAC output inside CAF, use '-c:a flac' and remove the '-b:a' flag entirely, since FLAC is a lossless codec that doesn't use bitrate targeting. For example: 'ffmpeg -i input.flv -vn -c:a flac output.caf'. CAF supports all these codecs natively, giving you flexibility depending on whether your target application needs compressed or uncompressed audio.
Yes. On macOS or Linux you can run a shell loop: 'for f in *.flv; do ffmpeg -i "$f" -vn -c:a pcm_s16le -b:a 128k "${f%.flv}.caf"; done'. On Windows PowerShell, use: 'Get-ChildItem *.flv | ForEach-Object { ffmpeg -i $_.FullName -vn -c:a pcm_s16le -b:a 128k ($_.BaseName + ".caf") }'. This is especially practical for bulk-archiving large libraries of legacy Flash video where only the audio is needed.
CAF is an Apple-designed format and has its strongest support within the Apple ecosystem — macOS, iOS, Logic Pro, GarageBand, and Core Audio. It is not natively supported by most Windows or Android applications, and even some professional cross-platform DAWs treat it as a secondary format. If you need to use the extracted audio outside of Apple tools, consider converting the FLV audio to WAV (pcm_s16le) or FLAC instead, as those formats have universal compatibility while still offering uncompressed quality.
FLV's metadata support is minimal — it can carry basic stream properties but rarely holds standard ID3-style tags for music metadata. During conversion to CAF, FFmpeg will attempt to pass through any metadata it finds, but in practice most FLV files contain little to no meaningful tag data, so the CAF output will typically have an empty or near-empty metadata block. If you need to embed title, artist, or album tags in the output CAF, you can append FFmpeg metadata flags to the command, such as '-metadata title="My Track"'.
Technical Notes
FLV containers use the AAC audio codec at bitrates typically between 64k and 192k, or occasionally MP3 via libmp3lame — both lossy formats. When demuxed and decoded to pcm_s16le for CAF output, the resulting file will be significantly larger than the source FLV; a 10-minute mono 44.1kHz AAC track at 128k might produce a ~50MB uncompressed CAF file versus a ~10MB FLV. CAF's 64-bit chunk size architecture means this expansion is never a container-level problem, even for very long recordings. The -b:a 128k flag in the command has no meaningful effect when the output codec is pcm_s16le, since PCM is an uncompressed format whose bitrate is determined entirely by sample rate and bit depth — it remains in the command for structural consistency with the tool's interface. FLV does not support embedded subtitle tracks or chapter markers, so no metadata of that kind will be lost in the conversion. One known limitation: FLV files with variable or non-standard sample rates (e.g., 11025 Hz from older Flash encoders) will be preserved as-is in the CAF output, but some Apple audio applications may resample them on import.