Extract Audio from CAVS to AAC — Free Online Tool
Extract the AAC audio track from a CAVS (Chinese Audio Video Standard) file and save it as a standalone .aac file. Since CAVS files already carry AAC audio, this conversion remuxes the audio stream directly without re-encoding, preserving the original quality at near-instant speed.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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Settings
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 encode video using a codec related to AVS1 (China's national standard) and package audio as AAC. This tool strips the video stream entirely using the -vn flag and copies the existing AAC audio bitstream directly into a bare .aac container. Because the audio codec in the source (AAC) matches the output format (AAC), no re-encoding is necessary by default — the raw audio data is extracted and rewrapped without any quality loss or transcoding overhead. If you choose a different bitrate in the tool's options, FFmpeg will re-encode the audio to match, which does introduce a small lossy generation. At the default 128k setting matching the source, extraction is essentially lossless in practice.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application, the open-source multimedia processing engine that handles demuxing the CAVS container, stream selection, and writing the AAC output file. |
-i input.cavs
|
Specifies the input CAVS file. FFmpeg reads and demuxes the CAVS container, identifying the video stream (AVS1/CAVS codec) and the audio stream (AAC) so they can be handled separately. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the CAVS video stream and produce an audio-only output. Without this flag, FFmpeg would attempt to include video in the output, which the bare .aac container cannot hold. |
-c:a aac
|
Sets the audio codec to AAC. Since the source CAVS file already contains AAC audio, this either copies the stream directly or re-encodes it using FFmpeg's built-in AAC encoder if the bitrate differs from the source. |
-b:a 128k
|
Sets the target audio bitrate to 128 kilobits per second, a standard quality level for AAC that balances file size and audio fidelity suitable for music and broadcast speech extracted from CAVS recordings. |
output.aac
|
Specifies the output filename with the .aac extension, directing FFmpeg to write a raw ADTS-framed AAC bitstream — the most universally compatible format for standalone AAC audio files extracted from CAVS sources. |
Common Use Cases
- Extracting Chinese broadcast or IPTV audio captured in CAVS format to create an audio-only archive for archival or transcription purposes
- Pulling the audio commentary or narration track from a CAVS-encoded educational or government broadcast video to use in a podcast or audio-only presentation
- Extracting the AAC audio from a CAVS file to load into mobile apps or iTunes-compatible players that cannot handle the CAVS video container
- Isolating the audio stream from CAVS broadcast recordings for use in video editing timelines where only the audio reference is needed
- Converting a large CAVS broadcast file into a small AAC audio file to reduce storage when the video content is no longer needed
- Preparing AAC audio extracted from CAVS source material for streaming or web playback on platforms that do not support the CAVS container
Frequently Asked Questions
At the default 128k setting, no additional quality loss occurs because the tool copies the existing AAC bitstream directly out of the CAVS container without re-encoding it. The AAC audio was already compressed when the CAVS file was created, so that original lossy generation remains, but no new compression artifacts are introduced. Only if you select a different bitrate will FFmpeg re-encode the audio, adding a second generation of lossy compression.
A raw .aac file is an ADTS-framed AAC bitstream without any container wrapper, which is the simplest and most direct output when you only want the audio. An .m4a file uses the MPEG-4 container and adds metadata support and better seeking, but requires additional muxing. For maximum compatibility with players like iTunes, iOS, and Android, either format works fine — but if you need chapter markers or rich metadata, consider remuxing the output .aac into an .m4a using a separate FFmpeg step.
Yes — replace '128k' in '-b:a 128k' with any supported value such as 64k, 192k, or 320k. If you raise the bitrate above what the source CAVS file uses, FFmpeg will re-encode the audio but cannot recover detail that was already discarded during original CAVS encoding, so the file gets larger without meaningful quality improvement. Lowering the bitrate (e.g., to 64k) will re-encode at a smaller size with noticeably reduced quality, which is useful for voice-only content where bandwidth matters.
CAVS (Chinese Audio Video Standard, also known as AVS) is a Chinese national video compression standard developed as an alternative to H.264/AVC, mandated for certain broadcast and government applications in China. CAVS files typically originate from Chinese digital television broadcasts, IPTV streams, or media produced under Chinese regulatory requirements. The audio track in these files is most commonly encoded as AAC, which is why extracting to .aac is a natural and lossless-path conversion.
Yes. AAC is natively supported on all Apple devices and platforms including iPhone, iPad, Mac, and iTunes/Apple Music. It is also compatible with Android, major browsers via HTML5 audio, Spotify ingestion pipelines, and most streaming services. The raw .aac ADTS format has slightly less metadata support than .m4a, but playback compatibility is excellent across all modern players.
On Linux or macOS, you can loop over all CAVS files in a directory with: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a aac -b:a 128k "${f%.cavs}.aac"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a aac -b:a 128k "%~nf.aac". Each file is processed sequentially, extracting the AAC audio track and naming the output after the original file. The browser-based tool processes one file at a time, so the FFmpeg command approach is recommended for batch workflows or files over 1GB.
Technical Notes
CAVS (AVS1) is a relatively niche container primarily encountered in Chinese broadcast and IPTV contexts, and FFmpeg's support for it is read-focused — demuxing and audio extraction work reliably, but writing CAVS video is less common. The audio track in CAVS files is standardly AAC encoded at typical broadcast bitrates (often 128k–192k), making extraction to .aac straightforward with no codec translation needed. The output is a raw ADTS-framed AAC file, which lacks a proper container header for rich metadata such as title, artist, or album tags — these are better stored in an M4A (MPEG-4 Audio) wrapper. If the source CAVS file contains multiple audio tracks (uncommon but possible in broadcast recordings), FFmpeg will default to the first audio stream; use '-map 0:a:1' to select a different track. Files processed in the browser via FFmpeg.wasm are subject to available RAM; for very large CAVS broadcast recordings, running the command locally on your desktop is recommended and will be significantly faster.