Extract Audio from CAVS to AC3 — Free Online Tool
Extract the AAC audio track from a CAVS video file and convert it to AC3 (Dolby Digital) format. This tool transcodes the audio from CAVS's native AAC stream into AC3's bitstream structure, making it compatible with DVD authoring tools, Blu-ray players, AV receivers, and broadcast systems that require Dolby Digital audio.
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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 carry AAC-encoded audio alongside an H.264 video stream. This tool discards the video entirely using the -vn flag and re-encodes only the audio — transcoding from AAC to AC3 (the codec behind Dolby Digital). Because AAC and AC3 are both lossy formats with different encoding architectures, this is a full re-encode, not a remux. FFmpeg decodes the AAC audio from the CAVS container, then re-encodes it using the ac3 encoder at 192k bitrate by default. The output is a raw .ac3 bitstream file, which can be muxed into DVD VOB files, Blu-ray M2TS containers, or broadcast transport streams.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool. In the browser, this runs via FFmpeg.wasm — a WebAssembly port that executes entirely client-side without sending your CAVS file to any server. |
-i input.cavs
|
Specifies the input CAVS file. FFmpeg uses the cavs demuxer to read the container, identifying the H.264 video stream and the AAC audio stream embedded within the Chinese standard format. |
-vn
|
Disables video output entirely. This tells FFmpeg to ignore the H.264 video track from the CAVS file, ensuring only the audio stream is processed and no video data is written to the AC3 output. |
-c:a ac3
|
Selects the AC3 encoder for the audio stream, transcoding the source AAC audio into Dolby Digital format. This is a full re-encode — the AAC bitstream is decoded and then re-encoded using the ATSC A/52 AC3 codec. |
-b:a 192k
|
Sets the AC3 audio bitrate to 192 kilobits per second, which is a standard and widely compatible bitrate for Dolby Digital stereo audio in DVD and broadcast contexts. Higher values like 320k or 384k can be used to reduce generation loss from the AAC-to-AC3 transcode. |
output.ac3
|
Specifies the output filename with the .ac3 extension, producing a raw Dolby Digital bitstream file. This format is directly usable in DVD authoring, Blu-ray muxing, and broadcast encoding workflows. |
Common Use Cases
- Extracting audio from Chinese broadcast CAVS recordings for use in DVD authoring software that requires Dolby Digital AC3 audio tracks
- Preparing audio from CAVS source material for integration into AV receiver systems that natively decode AC3 but cannot handle AAC streams
- Converting the audio track from a CAVS video file for inclusion in a Blu-ray project where AC3 is the required or preferred audio codec
- Stripping and re-encoding audio from Chinese standard broadcast captures to AC3 for compatibility with home theater equipment and legacy set-top boxes
- Creating a standalone AC3 file from a CAVS recording for use in post-production workflows that require Dolby Digital formatted audio deliverables
- Testing audio quality and channel configuration from a CAVS broadcast capture before committing to a full video transcode pipeline
Frequently Asked Questions
Yes, some quality loss is expected because this conversion involves decoding a lossy AAC stream and re-encoding it into another lossy format, AC3. This generation loss is most audible at lower bitrates. To minimize degradation, use a higher AC3 bitrate such as 320k or 384k, especially if the source CAVS file was encoded at 192k AAC or above. At the default 192k AC3 setting, the result is generally transparent for most listening purposes.
The .ac3 extension denotes a raw Dolby Digital bitstream file containing only the audio data with no container wrapper. This is the standard bare format for AC3 and is directly compatible with DVD authoring tools, Blu-ray muxers like MakeMKV or tsMuxeR, and broadcast encoding systems. If you need the AC3 audio inside a container like MKV or MP4, you would mux the .ac3 file into that container as a separate step.
AC3 supports up to 5.1 surround sound (six channels), which is one of its defining strengths for home theater and broadcast use. However, CAVS files — particularly from Chinese broadcast sources — typically carry stereo AAC audio. FFmpeg will encode whatever channel configuration is present in the source, so a stereo CAVS source will produce a stereo AC3 file. You would need additional FFmpeg upmixing flags if you want to expand to 5.1.
Modify the -b:a flag value in the command. AC3 supports standard bitrates including 96k, 128k, 192k, 256k, 320k, 384k, 448k, and 640k. For example, to produce a higher-quality output, use: ffmpeg -i input.cavs -vn -c:a ac3 -b:a 320k output.ac3. For DVD compatibility, 192k is a common and well-supported target. For Blu-ray or high-fidelity applications, 384k or 448k is recommended.
A raw .ac3 bitstream file has no container-level metadata support, so any title, language, or artist tags embedded in the CAVS file will not be present in the output. AC3 bitstream metadata such as Dolby dialog normalization (dialnorm) values are set during encoding. If metadata preservation is important, consider muxing the AC3 into an MKV container using ffmpeg with the -map_metadata flag, which supports rich metadata.
Yes. On Linux or macOS, you can use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a ac3 -b:a 192k "${f%.cavs}.ac3"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a ac3 -b:a 192k "%~nf.ac3". The browser-based tool processes one file at a time, but the FFmpeg command is ideal for batch processing large collections locally.
Technical Notes
CAVS (Chinese Audio Video Standard) containers typically use H.264 video with AAC audio at 128k bitrate, reflecting its origins as a Chinese national broadcast standard. When extracting audio for AC3 conversion, the key consideration is that AAC and AC3 use fundamentally different psychoacoustic models and entropy coding schemes — AAC generally achieves better perceptual quality per kilobit than AC3, meaning a 128k AAC source transcoded to 192k AC3 will not improve quality, but rather change the encoding architecture to one that is more broadly compatible with legacy and professional broadcast equipment. The -vn flag is essential here to prevent FFmpeg from attempting to encode or copy the H.264 video stream, which has no place in a raw .ac3 output file. The resulting AC3 file uses the ATSC A/52 standard bitstream format. Note that CAVS files are relatively uncommon outside of China, and some FFmpeg builds may require the cavs demuxer to be compiled in; the standard FFmpeg release includes this support. AC3's fixed bitrate encoding and standard frame structure make it highly predictable for downstream muxing into DVD or broadcast transport stream containers.