Convert CAVS to FLAC — Free Online Tool
Extract and losslessly archive the audio track from a CAVS video file as FLAC, preserving every bit of audio detail using the FLAC codec's lossless compression. Ideal for archiving audio from Chinese broadcast or standard-compliant CAVS media without any quality degradation.
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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 a video stream (typically encoded with a CAVS-specific codec) and an AAC audio track. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio from the CAVS container. That decoded audio is then re-encoded using the FLAC codec at compression level 5, which is lossless — meaning the output FLAC file is a bit-perfect representation of the original audio signal, just stored more efficiently. Because AAC is a lossy format, the FLAC output preserves exactly what was in the AAC stream, but cannot recover audio detail that AAC's lossy encoding originally discarded. The result is a losslessly compressed audio file with no further generation loss.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles all demuxing, decoding, encoding, and muxing operations in this conversion. |
-i input.cavs
|
Specifies the input file in CAVS format. FFmpeg detects the CAVS container, identifies the video and AAC audio streams inside, and makes them available for processing. |
-c:a flac
|
Instructs FFmpeg to encode the audio stream using the FLAC codec. This produces a losslessly compressed audio output — the decoded AAC signal is re-encoded to FLAC with zero additional quality loss. |
-compression_level 5
|
Sets FLAC's compression effort to level 5 out of 8. This is the FLAC default and balances encoding speed against output file size; all compression levels produce bit-identical audio, so only storage size and processing time are affected. |
output.flac
|
Defines the output file path and name. The .flac extension tells FFmpeg to wrap the encoded FLAC audio stream in a standard FLAC file container, which is compatible with virtually all audio players and archiving tools. |
Common Use Cases
- Archiving the audio commentary or dialogue track from CAVS broadcast recordings in a lossless format for long-term storage
- Extracting a music or soundtrack audio track from a CAVS video file to import into a DAW or audio editing application without introducing additional lossy compression
- Preparing audio from Chinese national standard CAVS media for quality comparison or forensic analysis, where a stable lossless container is required
- Converting CAVS audio to FLAC so it can be played or indexed by music library software (e.g., Plex, Foobar2000) that does not support CAVS container playback
- Creating a FLAC master archive of broadcast audio before performing further lossy exports (e.g., MP3 or Opus) to avoid cumulative quality loss across multiple encodes
Frequently Asked Questions
No. The FLAC codec is lossless, meaning it introduces zero additional quality loss, but it cannot restore detail that AAC's original lossy encoding already discarded. The FLAC file will be a bit-perfect preservation of the decoded AAC signal — identical in quality to what you hear in the CAVS file, but not better. Think of it as freezing the current quality permanently rather than improving it.
The AAC audio track inside a CAVS file is highly compressed using lossy encoding, which achieves very small file sizes by permanently discarding audio data. FLAC uses lossless compression, which must retain all audio data, so its files are inherently larger than AAC at equivalent durations. At compression level 5, FLAC balances encode speed and file size well, but it will still typically be several times larger than a 128k AAC stream.
CAVS containers carry limited metadata compared to formats like MKV or MP4, and FFmpeg will attempt to map any available metadata tags to the FLAC output's Vorbis comment fields. However, CAVS is primarily a broadcast format and may contain little or no embedded metadata beyond stream properties. You may need to tag the resulting FLAC file manually using a tool like MusicBrainz Picard or a FLAC tag editor.
Replace the value after -compression_level in the command. Valid values range from 0 (fastest encoding, largest file) to 8 (slowest encoding, smallest file). For example, use -compression_level 8 for maximum compression or -compression_level 0 for the fastest possible encode. Crucially, all levels produce bit-identical audio output — only the file size and encoding speed differ, never the audio quality.
Yes. On Linux or macOS, you can use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -c:a flac -compression_level 5 "${f%.cavs}.flac"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -c:a flac -compression_level 5 "%~nf.flac". Each CAVS file will be processed sequentially, producing a separate FLAC file per input.
No. CAVS does not support embedded chapters, and FLAC's cue sheet feature requires manual creation — it is not populated automatically during this conversion. If your CAVS content has time-based segment information encoded externally, you would need to author a FLAC cue sheet separately after conversion.
Technical Notes
CAVS (Chinese Audio Video Standard) is a broadcast-oriented container and codec standard developed in China, and FFmpeg's support for it covers demuxing and audio extraction reliably, though the format sees limited use outside Chinese broadcast infrastructure. The audio track in CAVS files is almost always AAC, encoded at typical broadcast bitrates (commonly 128k or 192k). When FFmpeg extracts this audio and encodes to FLAC, the PCM intermediate is losslessly captured — no resampling or channel remapping occurs unless the source has an unusual configuration. FLAC at compression level 5 (the default) offers a solid balance between encode time and output file size, typically achieving 50–60% of uncompressed PCM size for mixed audio content. One key limitation: because the source AAC is lossy, audiophile use cases expecting CD-quality or better lossless audio will not be served by this workflow — the FLAC simply locks in whatever fidelity the AAC encoding preserved. FLAC's seeking and ReplayGain features are fully available in the output and can be added post-conversion with standard FLAC tools.