Extract Audio from CAVS to MP3 — Free Online Tool
Extract the AAC audio track from a CAVS (Chinese Audio Video Standard) video file and convert it to MP3 using the LAME encoder. This tool re-encodes the audio from AAC to libmp3lame in your browser — no uploads required — producing a universally compatible MP3 file from Chinese broadcast or digital media content.
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 contain an AAC audio track paired with an AVS video stream. Since MP3 is an audio-only format, the video stream is completely discarded using the -vn flag. The AAC audio cannot simply be remuxed into MP3 — it must be fully decoded and then re-encoded using the LAME MP3 encoder (libmp3lame), introducing a second generation of lossy compression. The output is a standalone MP3 file encoded at 128k bitrate by default, which is compatible with virtually every audio player, device, and platform in existence.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, the open-source multimedia processing engine that handles decoding the CAVS container and its AVS video and AAC audio streams, then re-encoding the audio as MP3. |
-i input.cavs
|
Specifies the input CAVS file. FFmpeg will demux this file, identifying the AVS video stream and the AAC audio stream contained within the Chinese Audio Video Standard container. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore and discard the AVS video stream from the CAVS file. Since the target is MP3 (an audio-only format), no video data is needed or allowed in the output. |
-c:a libmp3lame
|
Selects the LAME MP3 encoder to re-encode the AAC audio from the CAVS file into MPEG Layer III (MP3) format. The AAC-to-MP3 conversion requires full decode-and-reencode since the two codecs are incompatible for stream copying. |
-b:a 128k
|
Sets the MP3 output audio bitrate to 128 kilobits per second, a widely accepted default that balances file size and perceptual audio quality. This can be increased to 192k, 256k, or 320k in the command if the CAVS source contained high-bitrate AAC audio worth preserving more faithfully. |
output.mp3
|
Defines the output filename and tells FFmpeg to write an MP3 file. The .mp3 extension signals FFmpeg to use the MP3 muxer, packaging the libmp3lame-encoded audio into a standard MP3 file compatible with virtually every audio player and platform. |
Common Use Cases
- Extracting dialogue, commentary, or narration from CAVS-encoded Chinese broadcast recordings for transcription or subtitling work
- Pulling the audio from CAVS media files obtained from Chinese digital TV or VOD services to create portable audio files playable on any device
- Archiving the audio portion of CAVS content when the video is no longer needed but the spoken content or soundtrack has lasting value
- Preparing audio from CAVS source material for upload to podcast platforms or audio streaming services that require MP3 format
- Converting CAVS broadcast recordings to MP3 so the audio can be edited in standard DAWs or audio editors that may not support CAVS containers
- Extracting music or ambient audio from CAVS-format Chinese film or television content for personal listening without carrying the large video file
Frequently Asked Questions
Yes, some quality loss is unavoidable. The original audio in a CAVS file is already AAC-encoded (lossy), and converting it to MP3 requires decoding the AAC and re-encoding with the LAME encoder — a second lossy compression step. At 128k MP3 bitrate, the result is generally acceptable for speech and most music, but audiophiles working with high-quality source material may want to increase the bitrate to 192k or 320k in the FFmpeg command to minimize the quality gap.
MP3 is a format that only supports MPEG Layer III audio data — it cannot contain AAC-encoded audio streams. The MP3 container and codec are inseparable, so any non-MP3 audio source, including the AAC track in a CAVS file, must be fully decoded and re-encoded as MP3. There is no lossless copy path from CAVS to MP3.
CAVS files typically carry minimal metadata compared to formats like MP4 or MKV, and the FFmpeg command shown here does not include explicit metadata mapping flags. FFmpeg will attempt to transfer any available title or comment metadata, but fields like artist, album, and track number are unlikely to be populated in the output MP3. You will generally need to add ID3 tags to the resulting MP3 manually using a tag editor after conversion.
Replace the value after -b:a in the command with your desired bitrate. For example, to encode at 320k (the highest standard MP3 bitrate), use: ffmpeg -i input.cavs -vn -c:a libmp3lame -b:a 320k output.mp3. For most speech content from broadcast CAVS sources, 128k is sufficient, but 192k or 256k is a good compromise for music or high-fidelity audio where the source AAC was encoded at a high bitrate.
Yes. On Linux or macOS, you can use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a libmp3lame -b:a 128k "${f%.cavs}.mp3"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a libmp3lame -b:a 128k "%~nf.mp3". This is especially useful when processing large collections of CAVS recordings from Chinese broadcast archives. The browser-based tool handles one file at a time.
MP3 is the best choice when maximum compatibility is the priority — it plays on every device, car stereo, podcast platform, and audio editor without exception. However, if you want to minimize the double-lossy-compression penalty, consider extracting to AAC (M4A) instead, which would allow a direct stream copy of the existing audio without re-encoding. MP3 is ideal for sharing, archiving for playback, or uploading to platforms; AAC copy would be better for preserving original audio quality when re-encoding is undesirable.
Technical Notes
CAVS (Chinese Audio Video Standard) is a Chinese national broadcast standard primarily used in domestic Chinese digital television and media distribution. The audio track in CAVS files is encoded in AAC, which is itself a modern, efficient lossy codec — meaning the source audio entering this conversion has already been compressed once. The MP3 output uses the libmp3lame encoder, the gold standard open-source LAME implementation, at a default bitrate of 128k. CAVS does not support multiple audio tracks, subtitles, or chapters, so there is no risk of losing those elements — the format simply doesn't carry them. The -vn flag ensures the AVS video stream is completely ignored and not processed, which significantly speeds up conversion since no video decoding occurs. File sizes for the resulting MP3 will be substantially smaller than the CAVS source, as only the audio bitstream is retained. Note that CAVS container support in FFmpeg depends on the build; most standard FFmpeg builds include AVS/CAVS demuxing support, but some minimal or stripped builds may not, which would produce a 'no such demuxer' error when running the command locally.