Extract Audio from CAVS to AIFC — Free Online Tool
Extract audio from CAVS (Chinese Audio Video Standard) video files and save it as AIFC, Apple's compressed Audio Interchange File Format. This tool decodes the AAC audio track embedded in your CAVS container and re-encodes it to 16-bit big-endian PCM (pcm_s16be) inside an AIFC wrapper — ideal for professional audio workflows that require uncompressed or lightly compressed archival-grade audio.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your CAVS file here
or click to browse
Free — no uploads, no signups. Your files never leave your browser.
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 store their audio as AAC — a lossy compressed codec. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio to raw PCM samples. Those samples are then written as 16-bit signed big-endian PCM (pcm_s16be) into an AIFC container. AIFC is an extension of Apple's AIFF format that supports both uncompressed PCM and compressed variants; pcm_s16be is uncompressed, so the output is lossless from the point of decoding — though any quality loss from the original AAC encoding in the CAVS file cannot be recovered. The resulting AIFC file is substantially larger than the CAVS audio track but is fully uncompressed and suitable for professional editing environments.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg multimedia processing engine — the same underlying library that powers this browser-based tool via WebAssembly. Running this command locally on your desktop will produce identical output to the browser tool. |
-i input.cavs
|
Specifies the input file — a CAVS-formatted video container, which typically carries a libx264 video stream and an AAC audio stream. FFmpeg reads both streams but only the audio will be used in this conversion. |
-vn
|
Disables video output entirely, stripping the libx264 video track from the CAVS file. This ensures the output AIFC file contains only the extracted audio with no video data. |
-c:a pcm_s16be
|
Sets the audio codec to 16-bit signed big-endian PCM — the native uncompressed audio format for AIFC/AIFF containers. This decodes the lossy AAC from the CAVS source into fully uncompressed PCM samples, which are then written verbatim into the AIFC file. |
-b:a 128k
|
Specifies a target audio bitrate of 128 kbps. For uncompressed PCM codecs like pcm_s16be, this value has no effect since the bitrate is mathematically fixed by sample rate, bit depth, and channel count — but the flag is harmless and included for consistency with the tool's interface. |
output.aifc
|
Names the output file and signals to FFmpeg that the container format should be AIFC. FFmpeg infers the AIFF/AIFC muxer from the .aifc extension, wrapping the pcm_s16be audio stream in a standards-compliant Apple AIFC file. |
Common Use Cases
- Extracting the audio track from a CAVS broadcast recording for import into a professional DAW like Pro Tools or Logic Pro, which natively support AIFC/AIFF files.
- Archiving audio from Chinese broadcast or IPTV CAVS streams in an uncompressed PCM format for long-term preservation without further lossy compression generations.
- Preparing audio sourced from CAVS video content for use in Apple ecosystem workflows, where AIFC is a standard interchange format in Final Cut Pro and older QuickTime-based pipelines.
- Removing the video layer from a CAVS file to produce a clean, uncompressed audio asset for music or dialogue re-use in post-production.
- Converting CAVS news broadcast audio into AIFC for compliance with broadcasters or archives that mandate uncompressed audio deliverables.
- Isolating audio from CAVS-encoded content recorded by Chinese digital TV receivers for analysis, transcription, or subtitling workflows.
Frequently Asked Questions
No — the output quality is bounded by the AAC encoding that was applied when the CAVS file was originally created. AAC is a lossy codec, so some audio information was discarded at that stage. What this conversion does is decode the AAC stream and store it uncompressed in AIFC (pcm_s16be), preventing any further quality degradation. The result is a lossless representation of whatever quality the AAC track contained, not a restoration of the original pre-AAC audio.
The CAVS file stores audio as compressed AAC, which achieves significant file size reduction through lossy compression. The AIFC output stores the decoded audio as uncompressed 16-bit PCM at its full sample rate, with no compression applied. For a stereo audio track at 44.1 kHz, uncompressed PCM requires roughly 10 MB per minute — compared to around 1 MB per minute at 128k AAC. The video stream is removed, but the remaining audio is substantially larger in raw PCM form.
Yes. AIFC with pcm_s16be encoding is a well-supported format in Apple's professional audio and video tools. Final Cut Pro, Logic Pro, and GarageBand all handle AIFC files natively. The big-endian PCM encoding used here is the historical standard for AIFF/AIFC on Apple platforms, so compatibility is excellent without any additional conversion steps.
To change the output codec, replace pcm_s16be in the command with another codec supported by AIFC — for example, pcm_s24be for 24-bit depth (common in professional audio), pcm_f32be for 32-bit float, or pcm_alaw for G.711 A-law compressed audio. For example: ffmpeg -i input.cavs -vn -c:a pcm_s24be output.aifc. Note that the -b:a bitrate flag is effectively ignored for uncompressed PCM codecs since bitrate is determined entirely by sample rate, bit depth, and channel count.
Metadata preservation from CAVS to AIFC is limited. CAVS containers store metadata in a format that does not map cleanly to AIFF/AIFC metadata chunks, so tags like title, artist, or language may be dropped during conversion. If metadata retention is important, inspect the output with a tool like ffprobe or MediaInfo after conversion, and consider adding tags manually using FFmpeg's -metadata flag.
Yes — on the command line, you can wrap the FFmpeg command in a shell loop. On Linux or macOS: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a pcm_s16be -b:a 128k "${f%.cavs}.aifc"; done. On Windows Command Prompt: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a pcm_s16be -b:a 128k "%~nf.aifc". The browser-based tool processes one file at a time, so the FFmpeg command is especially useful for batch jobs involving many CAVS files.
Technical Notes
CAVS (Chinese Audio Video Standard, GB/T 20090) is primarily encountered in Chinese broadcast, IPTV, and optical disc contexts. Its audio track is encoded as AAC, which is a standard lossy codec — meaning the audio in any CAVS file has already undergone one generation of lossy compression before this conversion begins. AIFC (Audio Interchange File Format Compressed) is a superset of Apple's AIFF format; despite the 'Compressed' name, it supports uncompressed PCM codecs including pcm_s16be (16-bit, big-endian, signed), which is what this tool uses by default. Big-endian byte ordering is the native format for AIFF/AIFC and is fully expected by Apple tools, though little-endian systems (Windows, most modern hardware) handle it transparently. The -b:a 128k flag in the command has no practical effect when using uncompressed PCM codecs like pcm_s16be, since PCM bitrate is a fixed function of sample rate × bit depth × channels — it is included for completeness but can be omitted. One known limitation: CAVS files encountered in the wild may occasionally have non-standard or absent audio stream metadata, which can cause FFmpeg to infer sample rate or channel count incorrectly; if the output sounds wrong, inspect the source with ffprobe to verify stream parameters. AIFC files produced here will not contain chapter markers or subtitle data, as neither CAVS nor AIFC supports those features in this workflow.