Convert CAVS to MP3 — Free Online Tool
Extract and convert audio from CAVS video files into universally compatible MP3 format. This tool strips the AAC audio track from Chinese Audio Video Standard files and re-encodes it using the LAME MP3 encoder, delivering a standalone audio file playable on virtually any device or platform.
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FFmpeg Command
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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 contain an AAC audio stream packaged alongside a video stream encoded with a proprietary Chinese standard codec. During this conversion, FFmpeg discards the video stream entirely and re-encodes only the AAC audio into MP3 using the libmp3lame encoder. Because AAC and MP3 are both lossy formats with different compression algorithms, this is a full transcode — not a remux — meaning the audio is decoded from AAC and then re-encoded into MP3. The default bitrate of 128k provides a reasonable balance between file size and perceptual quality, though there is a minor generational quality loss inherent in transcoding between two lossy codecs.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles decoding the CAVS container and its AAC audio stream, then re-encoding it as MP3. |
-i input.cavs
|
Specifies the input file in CAVS format. FFmpeg reads both the video and audio streams from this file, though only the audio will be used in the output. |
-c:a libmp3lame
|
Selects the libmp3lame encoder for the audio stream, which transcodes the decoded AAC audio from the CAVS file into MPEG-1 Audio Layer III (MP3) format using the industry-standard LAME library. |
-b:a 128k
|
Sets the MP3 audio bitrate to 128 kilobits per second, a common default that balances file size and audio quality. For CAVS broadcast content, this is generally sufficient for speech; increase to 192k or 320k for music or higher fidelity needs. |
output.mp3
|
Defines the output filename and format. The .mp3 extension tells FFmpeg to wrap the encoded audio in an MP3 file — a format universally supported by media players, browsers, and audio editors, with no video stream included. |
Common Use Cases
- Extracting a speech, lecture, or broadcast segment from a CAVS recording produced by Chinese broadcast or IPTV equipment for use in a podcast or audio archive
- Converting CAVS media files from Chinese digital TV set-top boxes into MP3 so the audio can be played on standard car stereos, MP3 players, or phones that do not support CAVS
- Pulling a music performance or concert recording stored in CAVS format into MP3 for upload to music sharing or streaming platforms
- Archiving the audio commentary or narration from a CAVS-encoded documentary or educational video into a portable format for offline listening
- Preparing audio from CAVS broadcast footage for import into audio editing software like Audacity or Adobe Audition, which may not natively handle CAVS container input
- Stripping the audio from CAVS security or surveillance footage captured by Chinese-standard recording hardware for use in evidence documentation or review
Frequently Asked Questions
Some quality loss is unavoidable because both AAC (used in CAVS) and MP3 are lossy formats. When you transcode from one lossy format to another, the audio is fully decoded and then re-compressed, introducing what is called generational loss. At the default 128k bitrate the result is generally acceptable for speech and casual listening, but if the original AAC audio was encoded at a low bitrate, artifacts may be more noticeable. For the best possible MP3 output, select the highest bitrate option (320k).
MP3 is a pure audio format — it has no container structure capable of holding a video stream. The FFmpeg command targets only the audio stream from your CAVS file and discards the video entirely by outputting to an MP3 container. If you want to preserve the video, you would need to convert to a video format like MP4 instead.
CAVS files rarely carry rich metadata, and the conversion command does not explicitly map metadata tags. MP3 supports ID3 tags for artist, title, album, and similar fields, but these will only be present in the output if the source CAVS file contained compatible metadata that FFmpeg can forward. In practice, most CAVS broadcast recordings will produce an MP3 with little or no embedded metadata, so you may need to tag the file manually after conversion.
Replace the value after -b:a with your desired bitrate. For example, to encode at 320k (the highest quality MP3 setting), the command becomes: ffmpeg -i input.cavs -c:a libmp3lame -b:a 320k output.mp3. Higher bitrates produce larger files but reduce compression artifacts. If file size is a priority and the content is speech-only, 96k is often sufficient.
Yes. On Linux or macOS you can run a shell loop: for f in *.cavs; do ffmpeg -i "$f" -c:a libmp3lame -b:a 128k "${f%.cavs}.mp3"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -c:a libmp3lame -b:a 128k "%~nf.mp3". This processes every CAVS file in the current directory and creates a corresponding MP3 file with the same base name.
CAVS playback support is limited outside of China. VLC Media Player is the most reliable cross-platform option, as it includes a CAVS decoder. FFmpeg itself can also play CAVS files via ffplay. Most standard media players on Windows, macOS, and mobile operating systems do not natively support CAVS, which is one of the main practical reasons to extract the audio into MP3.
Technical Notes
CAVS (Chinese Audio Video Standard, also known as AVS or GB/T 20090) uses H.264/AVC-class compression for video and typically pairs it with AAC audio at bitrates common in broadcast contexts. The CAVS container is not widely recognized outside Chinese hardware and software ecosystems, so extracting its audio as MP3 significantly broadens compatibility. The libmp3lame encoder used here is the industry-standard open-source LAME library, which produces MP3 files compliant with MPEG-1 Audio Layer III. Because the audio transcode goes from AAC to MP3 — two perceptually-coded lossy formats with different psychoacoustic models — there is inherent transcoding loss; neither format's compressed data can be losslessly converted to the other. The -b:a flag controls constant bitrate (CBR) encoding; libmp3lame also supports variable bitrate (VBR) via the -q:a flag (values 0–9, lower is better) if you run the command locally. Chapters, multiple audio tracks, and subtitle streams are not applicable here since MP3 supports none of these features and CAVS support for them is minimal. The output MP3 will be a standard stereo or mono file depending on the channel configuration of the source audio.