Extract Audio from CAVS to OGA — Free Online Tool
Extract audio from CAVS (Chinese Audio Video Standard) video files and save it as an OGA file using the Vorbis codec — entirely in your browser. This tool strips the video stream and re-encodes the AAC audio track from the CAVS container into an Ogg Vorbis stream, producing a compact, open-format audio file compatible with free and open-source media players.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your CAVS file here
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Free — no uploads, no signups. Your files never leave your browser.
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 carry AAC audio encoded within a Chinese broadcast-standard container. Because OGA uses the Ogg container and supports Vorbis, Opus, or FLAC — not AAC — the audio cannot simply be remuxed; it must be fully decoded and re-encoded. FFmpeg reads the AAC audio track from the CAVS input, decodes it to raw PCM, and then re-encodes it using the libvorbis encoder at a variable-bitrate quality level of 4 (roughly 128–160 kbps). The video stream is discarded entirely using the -vn flag, so the output is a pure audio-only OGA file with no video data.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which is running here as FFmpeg.wasm compiled to WebAssembly, executing entirely inside your browser without sending your CAVS file to any server. |
-i input.cavs
|
Specifies the input file in CAVS format. FFmpeg identifies the CAVS video stream and the accompanying AAC audio stream from this container for processing. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore the CAVS video stream and produce an audio-only output — essential for creating a clean OGA audio file with no embedded video data. |
-c:a libvorbis
|
Selects the libvorbis encoder to re-encode the decoded AAC audio from the CAVS file into Ogg Vorbis format, which is the default and most widely supported audio codec for the OGA container. |
-q:a 4
|
Sets the libvorbis variable bitrate quality to 4 on a scale of 0–10, targeting approximately 128–160 kbps output — a balance between file size and audio fidelity suitable for most extracted broadcast or video audio content. |
output.oga
|
Defines the output filename with the .oga extension, which signals to FFmpeg that the audio should be wrapped in an Ogg container — the standard packaging for Ogg Vorbis audio-only files. |
Common Use Cases
- Extracting dialogue, narration, or ambient audio from Chinese broadcast or streaming CAVS video content for use in audio editing workflows
- Converting the audio track of a CAVS-encoded documentary or television recording into OGA for archiving in an open, royalty-free format
- Preparing audio from CAVS source material for playback on Linux-based systems or media players (such as VLC or Rhythmbox) that handle Ogg Vorbis natively
- Stripping the audio from a CAVS broadcast recording to create a standalone audio file for transcription or subtitling work
- Extracting a music or soundtrack track from a CAVS video file to store as a high-quality Vorbis audio file in a media library
- Reducing file size by discarding the video stream from a large CAVS recording when only the audio content is needed for review or distribution
Frequently Asked Questions
Yes, there will be some generation loss because the audio undergoes a decode-then-re-encode cycle — the AAC stream from the CAVS file is fully decoded to raw PCM and then re-encoded using libvorbis. This is unavoidable since OGA does not support AAC, meaning a direct stream copy is impossible. At the default quality setting of -q:a 4, libvorbis targets roughly 128–160 kbps, which is transparent for most listeners, but if the original CAVS audio was already heavily compressed, any artifacts present will be retained and potentially compounded.
Yes — OGA supports FLAC, which is a lossless codec. However, this tool defaults to Vorbis encoding because the source AAC audio in CAVS files is already lossy; encoding it losslessly with FLAC would perfectly preserve the lossy AAC decode but would produce a much larger file without recovering any of the quality lost in the original encoding. To use FLAC, you can modify the FFmpeg command shown on this page to use '-c:a flac' instead of '-c:a libvorbis -q:a 4'.
Adjust the '-q:a' flag value, which controls libvorbis's variable bitrate quality scale. The scale runs from 0 (lowest quality, smallest file) to 10 (highest quality, largest file), with 4 as the default (approximately 128–160 kbps). For example, changing the command to 'ffmpeg -i input.cavs -vn -c:a libvorbis -q:a 7 output.oga' targets roughly 224–256 kbps, which is suitable for critical listening. Avoid using '-b:a' for fixed bitrate with libvorbis unless you specifically need constant bitrate output.
CAVS (Chinese Audio Video Standard) primarily standardizes the video compression layer, using a Chinese-developed codec comparable to H.264. For the audio track, CAVS commonly pairs with AAC (Advanced Audio Coding) because AAC was already a mature, widely-supported standard at the time CAVS was developed and offered good compression efficiency. Some CAVS content may also use other audio codecs depending on the broadcaster, but AAC is the most common pairing in practice.
FFmpeg will attempt to copy any metadata tags present in the CAVS container to the OGA output, and the Ogg container supports standard metadata through Vorbis comment tags (such as TITLE, ARTIST, and ALBUM). However, CAVS files used in broadcast contexts often carry minimal or no embedded metadata, so the OGA output may simply be empty of tags. You can add or edit tags post-conversion using a tool like EasyTag or MusicBrainz Picard, both of which support OGA files.
The single-file command shown on this page can be adapted for batch processing in a shell script. On Linux or macOS, you can run: 'for f in *.cavs; do ffmpeg -i "$f" -vn -c:a libvorbis -q:a 4 "${f%.cavs}.oga"; done'. On Windows Command Prompt, use: 'for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a libvorbis -q:a 4 "%~nf.oga"'. The in-browser tool on this page processes one file at a time, so the desktop FFmpeg command is the recommended approach for bulk conversions.
Technical Notes
CAVS is a niche format primarily encountered in Chinese broadcast and media production pipelines, and support outside of FFmpeg and a handful of specialized players is limited. The audio stream in CAVS files is virtually always AAC, which cannot be stream-copied into an OGA container — the Ogg specification does not permit AAC payloads, so a full transcode through libvorbis (or libopus or flac) is mandatory. The default libvorbis setting of -q:a 4 produces variable bitrate output in the 128–160 kbps range, which is widely considered perceptually transparent. OGA is an audio-only subset of the Ogg container and is well-supported by open-source players including VLC, MPV, and Audacious, though it has limited support on Apple platforms without third-party software. The OGA format supports Vorbis comment metadata and chapter markers, making it useful for longer audio extracts such as lectures or broadcasts where structural markers may be helpful. Note that CAVS does not support multiple audio tracks, so there is no ambiguity about which audio stream to extract.