Convert CAVS to OGG — Free Online Tool
Convert CAVS video files to OGG audio using the Vorbis codec — stripping the Chinese national standard video stream and extracting a high-quality, open-format audio file compatible with Linux media players, web browsers, and Xiph-based tools. The conversion runs entirely in your browser via FFmpeg.wasm, with no file uploads required.
to
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 (Chinese Audio Video Standard) is a container carrying video encoded with its proprietary codec alongside an audio stream, most commonly AAC. During this conversion, FFmpeg discards the CAVS video stream entirely and re-encodes the audio from AAC into Vorbis (libvorbis), wrapping the result in an OGG container. This is a full audio transcode — not a remux — because AAC audio cannot be natively stored in OGG. The Vorbis encoder uses variable bitrate (VBR) quality scaling, so the output file size and quality depend on the -q:a setting rather than a fixed bitrate target. The result is an open, royalty-free audio file that plays natively in Firefox, Chromium-based browsers, and most Linux media applications.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool — in this browser-based tool, it runs as FFmpeg.wasm compiled to WebAssembly, processing your CAVS file locally without any server upload. |
-i input.cavs
|
Specifies the input file in CAVS (Chinese Audio Video Standard) format. FFmpeg reads both the proprietary CAVS video stream and the AAC audio stream from this container. |
-c:a libvorbis
|
Instructs FFmpeg to encode the audio stream using the libvorbis encoder, producing Vorbis audio — the native and most widely supported codec for OGG containers. The original AAC audio from the CAVS file is fully decoded and re-encoded into Vorbis. |
-q:a 4
|
Sets the Vorbis variable bitrate quality level to 4 on a 0–10 scale, targeting approximately 128 kbps average — a balanced default that delivers good audio fidelity while keeping file sizes reasonable. Increase this value toward 7–10 for higher quality archival output. |
output.ogg
|
Defines the output filename and tells FFmpeg to write an OGG container. The .ogg extension causes FFmpeg to mux the Vorbis audio stream into the Xiph OGG format, with no video stream included in the output. |
Common Use Cases
- Extracting the audio commentary or soundtrack from a CAVS broadcast recording for use in open-source video editing timelines that prefer Vorbis over AAC.
- Converting CAVS audio content from Chinese digital television broadcasts into OGG for archival in a fully open, patent-free format.
- Preparing audio from CAVS source files for upload to platforms or tools (such as MediaWiki or certain game engines) that specifically require OGG/Vorbis.
- Stripping video from a CAVS file to produce a lightweight OGG audio file for embedding in HTML5 web pages alongside an Opus or MP3 fallback.
- Extracting spoken audio from CAVS-encoded educational or documentary content to create podcast-ready or transcription-ready audio files in an open format.
- Running a local FFmpeg command on large CAVS broadcast files (over 1GB) to batch-convert audio archives to OGG for long-term open-format storage.
Frequently Asked Questions
Yes, some generation loss occurs because this conversion transcodes from one lossy format (AAC) to another (Vorbis). The original CAVS file's audio is decoded from AAC and then re-encoded into Vorbis, which introduces a second round of compression artifacts. To minimize quality loss, use a higher -q:a value (such as 7 or 8) when running the FFmpeg command. At the default -q:a 4, the Vorbis output is roughly equivalent to 128–160 kbps and is perceptibly transparent for most audio content.
OGG is primarily an audio container — while it technically supports the Theora video codec, this tool is configured to produce a standard audio-only OGG/Vorbis file, which is the format's most common and compatible use case. The CAVS video codec is a proprietary Chinese standard with very limited decoder support outside of dedicated broadcast hardware, so carrying it into OGG would produce a file that almost nothing can play. Dropping the video stream and extracting clean audio is the practical and universally compatible outcome.
Replace the -q:a 4 value in the command with a number between 0 and 10, where higher values produce better quality and larger files. For example, ffmpeg -i input.cavs -c:a libvorbis -q:a 7 output.ogg will produce noticeably higher quality audio suitable for archival. For smaller files where quality is less critical, -q:a 2 or -q:a 3 is a reasonable choice. Unlike fixed-bitrate encoding, Vorbis VBR quality mode allocates more bits to complex audio passages automatically.
Yes — OGG supports Opus audio as well as Vorbis. To use the Opus codec, change the command to: ffmpeg -i input.cavs -c:a libopus -b:a 128k output.ogg. Note that Opus uses bitrate-based encoding (-b:a) rather than the quality scale (-q:a) used by Vorbis. Opus generally achieves better quality than Vorbis at equivalent bitrates, especially below 128k, but Vorbis has slightly broader compatibility with older media players and game engines.
FFmpeg will attempt to copy any metadata tags present in the CAVS source file into the OGG container, which natively supports rich Vorbis Comment metadata including title, artist, album, and track fields. However, CAVS files from broadcast sources often carry minimal or non-standard metadata, so the OGG output may have few or no tags populated. You can add metadata manually by appending flags such as -metadata title='My Title' to the FFmpeg command before the output filename.
The command shown processes a single file, but you can adapt it for batch processing using a shell loop. On Linux or macOS: for f in *.cavs; do ffmpeg -i "$f" -c:a libvorbis -q:a 4 "${f%.cavs}.ogg"; done. On Windows Command Prompt: for %f in (*.cavs) do ffmpeg -i "%f" -c:a libvorbis -q:a 4 "%~nf.ogg". This is especially useful for processing large CAVS broadcast archives that exceed the browser tool's 1GB file limit.
Technical Notes
CAVS files carry audio encoded as AAC in most real-world broadcast implementations, which means the conversion path is AAC decode → Vorbis encode — a full lossy transcode with no possibility of lossless passthrough in the OGG container (unless FLAC is substituted as the codec, which requires the source to be decoded to PCM first anyway). The Vorbis encoder (libvorbis) in FFmpeg uses a VBR quality ladder where -q:a 4 targets approximately 128 kbps on average, with values ranging from -q:a 0 (very low quality, ~64 kbps) to -q:a 10 (near-transparent, ~500 kbps). OGG supports multiple audio tracks and chapter markers in the container spec, but since CAVS sources rarely carry more than one audio stream and FFmpeg maps only the default stream by default, multi-track output requires explicit -map flags. The CAVS video codec itself has no meaningful presence outside Chinese broadcast infrastructure, so this conversion is almost always an audio extraction workflow rather than a multimedia format migration. File sizes will typically be smaller than the source CAVS file given that video is removed, though the audio portion may be slightly larger or smaller depending on the original AAC bitrate versus the target Vorbis quality setting.