Convert CAVS to AAC — Free Online Tool
Extract and convert the AAC audio track from a CAVS video file into a standalone AAC audio file. This tool strips the video stream entirely, delivering the compressed audio in the universally compatible .aac container — ideal for getting audio out of Chinese broadcast or IPTV recordings.
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
CAVS files carry an AAC audio stream alongside their video, encoded with H.264/AVC. During this conversion, FFmpeg discards the video stream entirely and re-encodes (or passes through) the audio as AAC at a target bitrate of 128k. Because the source audio is already AAC, the transcoding step is minimal — the audio is decoded and re-encoded at the specified bitrate, which means quality is preserved very closely as long as the output bitrate is not significantly lower than the source. No video data is written to the output file, so the result is a pure audio .aac file compatible with virtually all modern devices and media players.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool — the underlying engine used both in this browser-based converter (via FFmpeg.wasm) and when running the command locally on your desktop. |
-i input.cavs
|
Specifies the input file in CAVS format. FFmpeg reads both the H.264 video stream and the AAC audio stream from this container, though only the audio will be used in the output. |
-c:a aac
|
Instructs FFmpeg to encode the output audio using the AAC codec. Since the source audio in the CAVS file is already AAC, this re-encodes it at the specified bitrate into a clean, universally compatible AAC stream. |
-b:a 128k
|
Sets the audio bitrate to 128 kilobits per second, which is the standard default for AAC and delivers good quality suitable for speech, music, and broadcast content at a compact file size. |
output.aac
|
Defines the output filename and container. The .aac extension tells FFmpeg to write a raw AAC audio file, with no video stream — compatible with Apple devices, streaming platforms, and virtually all modern audio players. |
Common Use Cases
- Extracting commentary, narration, or dialogue from a CAVS broadcast recording for use in a podcast or audio project
- Pulling the audio track from a CAVS IPTV stream to archive a radio-style program in a compact, widely compatible format
- Separating audio from a CAVS video file to load it into a mobile music or podcast app that does not support CAVS playback
- Preparing audio from Chinese broadcast CAVS recordings for upload to streaming platforms like Apple Podcasts or Spotify, which require AAC or MP3
- Reducing file size by discarding the large video stream from a CAVS file when only the audio content is needed for review or editing
Frequently Asked Questions
There is a small quality loss because the AAC audio in the CAVS file is decoded and re-encoded into a new AAC file — this is a generation loss, not a remux. However, since both the source and output use AAC at 128k, the loss is minimal and essentially imperceptible for most listening purposes. If you want to match the source quality exactly, make sure the output bitrate in the command matches the source bitrate, or use a slightly higher value like 192k.
It is technically possible to stream-copy the AAC audio from a CAVS file using '-c:a copy', which would produce a truly lossless extraction. However, CAVS containers encapsulate audio in a way that may require re-muxing to ensure clean AAC output compatibility. The default command re-encodes at 128k for maximum compatibility, but you can substitute '-c:a copy' in the command if you want to skip re-encoding entirely and trust the source stream is clean.
Replace '128k' in the '-b:a 128k' flag with your desired bitrate. Common options are 96k for smaller files, 192k for higher fidelity, or 256k and 320k for near-transparent quality. For example, to get high-quality audio you would run: ffmpeg -i input.cavs -c:a aac -b:a 192k output.aac. Keep in mind that raising the bitrate above the original source bitrate will not recover any detail that was lost during initial encoding.
Yes. On Linux or macOS you can loop over files in a directory with a shell command like: for f in *.cavs; do ffmpeg -i "$f" -c:a aac -b:a 128k "${f%.cavs}.aac"; done. On Windows using Command Prompt you can use: for %f in (*.cavs) do ffmpeg -i "%f" -c:a aac -b:a 128k "%~nf.aac". This is especially useful for processing collections of CAVS broadcast recordings.
Yes. AAC is Apple's preferred audio format and is natively supported across all Apple devices and iTunes. The output .aac file produced by this tool will play directly in the Music app on iOS and macOS, and can be imported into iTunes or synced to an iPhone or iPad without any additional conversion.
CAVS files typically carry minimal metadata compared to formats like MP4 or MKV, and what metadata does exist may not map cleanly to the AAC container. Duration will be preserved, but tags such as title or artist from the CAVS file are unlikely to transfer. If you need proper metadata in the output AAC file, you can add it using FFmpeg's '-metadata' flags, for example: -metadata title="My Recording" in the command.
Technical Notes
CAVS (Chinese Audio Video Standard) is a format primarily encountered in Chinese broadcast and IPTV contexts. Its audio track is encoded with AAC, making audio extraction straightforward for FFmpeg. Because CAVS is not a widely supported container outside of China, many standard media players on Western platforms cannot open these files — converting the audio track to a standalone .aac file solves this compatibility gap immediately. The output AAC file uses MPEG-4 Audio packaging, which is the standard expected by streaming services, mobile platforms, and desktop players globally. Note that the output format does not support subtitles, chapters, or multiple audio tracks, so this conversion is strictly a single audio stream extraction. If the CAVS source file has multiple audio tracks, FFmpeg will select the default (usually the first) track; to target a specific track, add '-map 0:a:1' (or the relevant index) before the output filename in the command.