Extract Audio from CAVS to FLAC — Free Online Tool
Extract lossless FLAC audio from CAVS video files — preserving the full fidelity of the AAC audio track by re-encoding it into FLAC's lossless compression. Ideal for archiving audio from Chinese broadcast or standard CAVS content without any further quality degradation.
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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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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 an AAC audio track alongside a libx264 video stream. Since AAC is a lossy codec and FLAC is lossless, the conversion cannot be a simple stream copy — FFmpeg must fully decode the AAC audio and then re-encode it into FLAC. The video stream is discarded entirely using the -vn flag, so only the audio pipeline runs. The resulting FLAC file contains a bit-perfect reconstruction of the decoded AAC audio, meaning no additional generation loss is introduced beyond what already existed in the original AAC encoding. FLAC then applies lossless compression to reduce file size without touching any sample data.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles the full pipeline of reading the CAVS container, demuxing the AAC audio stream, decoding it to PCM, and re-encoding it as FLAC. |
-i input.cavs
|
Specifies the input CAVS file. FFmpeg reads the container, identifies the libx264 video stream and the AAC audio stream, and makes both available for processing — though only the audio stream will be used in this extraction. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore the libx264 video stream in the CAVS file. This ensures the output FLAC file contains only audio data and that no unnecessary video decoding is performed. |
-c:a flac
|
Selects the FLAC encoder for the audio stream. FFmpeg decodes the source AAC audio to raw PCM and then encodes those samples using FLAC's lossless compression, producing audio that is bit-perfect to the decoded waveform. |
-compression_level 5
|
Sets FLAC's compression effort to level 5 on a scale of 0–8. This is the standard balanced setting that produces a reasonably compact FLAC file without spending excessive encoding time — audio quality is identical regardless of which level is chosen. |
output.flac
|
Defines the output file name and format. The .flac extension tells FFmpeg to write a standard FLAC file containing the losslessly compressed audio extracted from the CAVS source. |
Common Use Cases
- Archiving the audio commentary or dialogue from CAVS broadcast recordings in a lossless format for long-term preservation
- Extracting music or sound effects from CAVS video produced under the Chinese national broadcast standard for use in audio editing software like Audacity or DaVinci Resolve
- Converting the audio from CAVS source material into FLAC so it can be imported into a DAW for mixing, where lossless source files are required
- Stripping audio from CAVS files received from Chinese broadcasters or studios to create standalone archival audio masters
- Preparing FLAC audio extracted from CAVS content for distribution on platforms that require lossless audio, such as Bandcamp or high-resolution audio stores
- Recovering clean audio from a CAVS recording when the video portion is damaged or no longer needed
Frequently Asked Questions
FLAC is a lossless format, but the audio in your CAVS file was already compressed with AAC, which is lossy. This conversion decodes the AAC audio and encodes it into FLAC without introducing any new loss — every sample decoded from the AAC stream is preserved exactly in the FLAC file. However, the original lossy AAC encoding's quality ceiling cannot be recovered; FLAC simply freezes the audio at its current decoded state. Think of it as lossless preservation of what was already encoded, not a quality upgrade.
CAVS files use AAC audio, which achieves very small file sizes through lossy compression that permanently discards audio data. FLAC preserves every sample of the decoded audio without discarding anything, then applies lossless compression — but lossless compression cannot match the size reduction of lossy AAC at typical bitrates like 128k. The FLAC output represents the full uncompressed waveform in compressed form, so it will almost always be larger than the AAC audio it was decoded from.
The AAC bitrate setting in the CAVS source only governed how the original audio was encoded into AAC — it does not carry over to FLAC. FLAC uses a compression level (set here to 5) that controls the encoding speed and file size, but never the audio quality, which is always bit-perfect. The decoded 128k AAC audio will be stored in FLAC at whatever file size lossless compression achieves for that decoded waveform.
Yes. The -compression_level flag accepts values from 0 to 8. Level 0 encodes fastest with the least compression (larger file), while level 8 takes the most time but produces the smallest possible lossless FLAC file. The default here is 5, which is a widely used balance. To use maximum compression, change the command to: ffmpeg -i input.cavs -vn -c:a flac -compression_level 8 output.flac. Crucially, no compression level affects audio quality — all levels produce bit-identical audio output.
FFmpeg will attempt to copy any metadata tags present in the CAVS container (such as title or artist fields) into the FLAC file's Vorbis comment metadata block. However, CAVS files originating from Chinese broadcast workflows often carry minimal or no standard metadata tags, so the FLAC output may arrive without rich tagging. You can add or edit tags in the FLAC file afterward using tools like Mp3tag, MusicBrainz Picard, or metaflac.
The command shown processes a single file, but you can batch process on the desktop with a simple shell loop. On Linux or macOS: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a flac -compression_level 5 "${f%.cavs}.flac"; done. On Windows Command Prompt: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a flac -compression_level 5 "%~nf.flac". The browser-based tool on this page processes one file at a time.
Technical Notes
CAVS (Chinese Audio Video Standard) uses AAC audio encoded at a configurable bitrate (commonly 128k by default) inside a container designed for Chinese broadcast distribution. Because AAC is a lossy transform codec, the conversion to FLAC involves a full decode-encode cycle: FFmpeg decodes the AAC bitstream to raw PCM samples, then the FLAC encoder losslessly compresses those PCM samples. No sample data is altered in the FLAC encoding step, making this a stable archival path. The -vn flag ensures the libx264 video stream is completely ignored, meaning no video decoding overhead occurs. FLAC supports Vorbis comment metadata, cue sheets, and ReplayGain tags — features absent from the CAVS container spec — so the FLAC output can be enriched with these after conversion. One known limitation: if the CAVS source contains multichannel audio (uncommon but possible in broadcast contexts), FFmpeg will carry all channels into the FLAC output, but the browser-based tool is optimized for standard stereo streams. Files larger than 1GB should be processed using the desktop FFmpeg command directly.