Extract Audio from CAVS to AIFF — Free Online Tool
Extract lossless AIFF audio from CAVS video files directly in your browser. This tool strips the AAC audio track from China's broadcast-standard CAVS container and converts it to uncompressed PCM (16-bit big-endian) AIFF — the professional-grade format preferred on macOS — with no quality loss from re-compression.
to
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 AAC-encoded audio alongside H.264 video. This conversion discards the video stream entirely and decodes the AAC audio, then re-encodes it into uncompressed 16-bit big-endian PCM stored in an AIFF container. Because AAC is a lossy format, the output AIFF is a lossless representation of the already-decoded audio — meaning no additional lossy compression is applied, but the original AAC encoding artifacts (if any) are preserved in the decoded signal. The result is a large, uncompressed AIFF file that is fully compatible with macOS audio tools, DAWs, and professional audio workflows.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, the open-source multimedia processing engine that this browser-based tool runs via WebAssembly. All subsequent flags and arguments are passed to FFmpeg to define the conversion. |
-i input.cavs
|
Specifies the input file in CAVS (Chinese Audio Video Standard) format. FFmpeg reads and demuxes this container to access its H.264 video stream and AAC audio stream separately. |
-vn
|
Disables video output entirely, instructing FFmpeg to skip the H.264 video stream from the CAVS file. This is essential for audio extraction — without it, FFmpeg would attempt to include video, which AIFF cannot contain. |
-c:a pcm_s16be
|
Sets the audio codec to signed 16-bit big-endian PCM, which is the standard uncompressed audio encoding for AIFF. The AAC audio from the CAVS file is decoded and re-encoded as raw PCM — no lossy compression is applied in this step. |
output.aiff
|
Defines the output filename and signals to FFmpeg that the result should be written as an AIFF container. FFmpeg uses the .aiff extension to select the correct muxer, which packages the pcm_s16be audio stream into Apple's Audio Interchange File Format structure. |
Common Use Cases
- Extracting dialogue or narration audio from CAVS broadcast recordings for subtitling or dubbing projects targeting Chinese-language content
- Pulling soundtrack or music beds from CAVS media files into a macOS DAW like Logic Pro or GarageBand for editing in AIFF format
- Archiving the audio track of a CAVS video in an uncompressed, long-term preservation format before the original file is edited or destroyed
- Preparing audio extracted from Chinese broadcast footage for mastering workflows that require uncompressed AIFF input
- Isolating voice-over or commentary tracks from CAVS training or educational videos for transcription or localization
- Converting CAVS audio to AIFF so it can be imported into Apple Final Cut Pro or other macOS video tools that prefer native AIFF over AAC
Frequently Asked Questions
No — the output will not exceed the quality of the original AAC audio encoded in the CAVS file. CAVS uses AAC at typically 128k, a lossy format, so any encoding artifacts from that original compression are already baked into the audio signal. Converting to AIFF simply decodes the AAC and stores the result as uncompressed PCM; it does not recover lost information. The benefit of AIFF here is that no further quality is lost during the extraction process itself.
AIFF stores audio as uncompressed PCM, meaning every sample is written as raw data with no compression. AAC (used in CAVS) achieves roughly 10:1 compression ratios compared to uncompressed audio. For example, a CAVS file with 60 minutes of 128k AAC audio might produce an AIFF file several hundred megabytes in size. This is expected and is the nature of lossless uncompressed formats — you are trading file size for maximum compatibility and zero further quality loss.
CAVS files, rooted in a Chinese broadcast standard, typically carry minimal embedded metadata, and the AIFF format's metadata support (via AIFF chunks) is also limited compared to formats like FLAC or MP4. In practice, most CAVS-to-AIFF conversions via FFmpeg will not carry over meaningful ID3-style tags. If metadata preservation is critical, you should embed tags separately using a tool like bwfmetaedit or an audio tagger after extraction.
Yes — the default command uses pcm_s16be (16-bit big-endian PCM), but AIFF also supports pcm_s24be, pcm_s32be, pcm_f32be, and pcm_f64be. You can change the codec flag in the FFmpeg command to '-c:a pcm_s24be' for 24-bit output, for instance. However, since the source audio was AAC at 128k, choosing a higher bit depth will increase file size without recovering any additional audio quality — the original lossy encoding is still the limiting factor.
You can adapt the displayed command into a shell loop. On Linux or macOS, run: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.cavs}.aiff"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.aiff". This processes each CAVS file in the current directory and outputs a matching AIFF file. The browser-based tool handles one file at a time, so the FFmpeg command is especially useful for bulk processing large collections.
AIFF was developed by Apple and is natively supported on macOS, but it is readable on Windows through most professional audio applications including Adobe Audition, FL Studio, and VLC. Standard Windows Media Player does not play AIFF natively, and some consumer tools may not recognize it. If broad Windows compatibility is a priority, you might consider WAV instead, which uses the same uncompressed PCM audio but in a little-endian container with wider Windows software support.
Technical Notes
CAVS (Chinese Audio Video Standard) is a broadcast-oriented container format standardized in China, using H.264 video and AAC audio. Because CAVS is not commonly encountered outside Chinese broadcast and media production contexts, desktop tools with native CAVS support are rare — FFmpeg is one of the few open-source tools that can reliably demux it. The extraction command uses '-vn' to completely ignore the H.264 video stream, avoiding any video decoding overhead. The audio is decoded from AAC and written as pcm_s16be — signed 16-bit big-endian PCM — which is AIFF's default and most compatible codec. Big-endian byte ordering is a legacy of AIFF's origins on Motorola 68k-based Macs and is handled transparently by all modern AIFF-compatible software. One known limitation is that CAVS files from certain broadcast sources may have non-standard timing or container quirks that cause FFmpeg to emit warnings about timestamps; in most cases, the audio output is still correct. If you encounter sync issues, adding '-avoid_negative_ts make_zero' before the output filename can help stabilize timestamps.