Convert CAVS to AIFF — Free Online Tool
Extract and convert audio from CAVS video files into AIFF, Apple's uncompressed lossless audio format. This tool strips the AAC audio track from a Chinese Audio Video Standard file and re-encodes it as 16-bit big-endian PCM, giving you a pristine, uncompressed AIFF file ready for macOS audio workflows.
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FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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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 their video stream. During this conversion, the video stream is discarded entirely — only the audio is extracted. That AAC audio, which is lossy-compressed, is then decoded back to raw PCM samples and re-encoded into AIFF's native 16-bit big-endian PCM format (pcm_s16be). Because AAC is a lossy format, the resulting AIFF will be lossless and uncompressed going forward, but it cannot recover any audio detail that AAC's original compression discarded. Think of it as 'freezing' the audio at its current quality level in a professional, uncompressed container.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, which handles all media decoding, processing, and encoding. In the browser-based version of this tool, this runs as FFmpeg.wasm — a WebAssembly build of FFmpeg that executes entirely within your browser without sending any files to a server. |
-i input.cavs
|
Specifies the input file, a CAVS-format video. FFmpeg reads and demuxes this container to access its internal audio stream (AAC) and video stream, of which only the audio will be used in this conversion. |
-c:a pcm_s16be
|
Sets the audio codec to pcm_s16be — 16-bit signed PCM with big-endian byte ordering. This is the standard uncompressed audio encoding used in AIFF files, decoding the lossy AAC audio from the CAVS source into a raw, uncompressed PCM representation at the same resolution as CD audio. |
output.aiff
|
Defines the output file name and format. The .aiff extension tells FFmpeg to wrap the pcm_s16be audio stream in an AIFF container — Apple's Audio Interchange File Format — which is natively supported across macOS, Logic Pro, Final Cut Pro, and most professional audio applications. |
Common Use Cases
- Extracting a music or speech audio track from a CAVS broadcast recording for editing in Logic Pro or GarageBand on macOS, where AIFF is a natively supported format
- Archiving the audio portion of a CAVS video from a Chinese broadcast or streaming source in an uncompressed format for long-term preservation
- Preparing CAVS-sourced audio for import into professional Digital Audio Workstations (DAWs) that prefer or require uncompressed PCM audio like AIFF over compressed AAC
- Isolating commentary, dialogue, or narration from a CAVS-encoded video file to use as a standalone audio asset in a post-production pipeline
- Converting CAVS audio to AIFF for compatibility with Apple hardware and software ecosystems, including Final Cut Pro timelines that use AIFF as a reference audio format
- Stripping and preserving audio from CAVS educational or training videos distributed in China for use in cross-platform audio analysis or transcription tools
Frequently Asked Questions
Yes, the AIFF will sound identical to what you hear in the CAVS file, but it cannot be better than that. The original CAVS audio was encoded with AAC, a lossy codec, which permanently removed some audio detail during its initial compression. This conversion decodes that AAC stream and stores it uncompressed in AIFF — no further quality is lost, but the AAC-introduced losses are already baked in. The AIFF output is essentially a lossless snapshot of the lossy AAC audio.
AIFF stores audio as raw, uncompressed PCM samples — every audio sample is written in full with no compression whatsoever. The AAC audio inside a CAVS file, by contrast, is typically compressed to around 128 kbps. A single minute of 16-bit stereo AIFF audio at 44.1 kHz consumes roughly 10 MB, while the same AAC audio might only be 1 MB. Additionally, the CAVS file included a video stream, but the AIFF output is audio-only, so it won't be larger than the original overall file — just larger than the audio portion alone.
The output uses pcm_s16be — 16-bit signed PCM in big-endian byte order, which is the standard AIFF audio encoding and the default Apple uses. If you need higher bit depth, you can modify the FFmpeg command to use pcm_s24be for 24-bit audio or pcm_s32be for 32-bit, depending on your downstream application's requirements. For most broadcast and general use cases, pcm_s16be is the appropriate choice and matches CD-quality audio resolution.
No. AIFF is a strictly audio-only container format — it has no capability to store video streams. FFmpeg automatically handles this by extracting only the audio stream from the CAVS input. The video data from the CAVS file is completely discarded during conversion, and no video-related quality settings apply to this process.
Replace 'pcm_s16be' in the command with another supported AIFF codec to change the bit depth. For example, use '-c:a pcm_s24be' for 24-bit audio, which is common in professional audio production, or '-c:a pcm_f32be' for 32-bit floating-point PCM if your DAW requires it. The full modified command would look like: ffmpeg -i input.cavs -c:a pcm_s24be output.aiff. Note that since AIFF is uncompressed, there is no bitrate parameter to tune — only the bit depth matters.
Yes. On Linux or macOS, you can run a shell loop: 'for f in *.cavs; do ffmpeg -i "$f" -c:a pcm_s16be "${f%.cavs}.aiff"; done'. On Windows Command Prompt, use: 'for %f in (*.cavs) do ffmpeg -i "%f" -c:a pcm_s16be "%~nf.aiff"'. This processes each CAVS file in the current directory and outputs a corresponding AIFF file, which is especially useful when archiving a collection of CAVS broadcast recordings.
Technical Notes
CAVS (Chinese Audio Video Standard) files typically encapsulate video encoded with the AVS or AVS2 codec alongside AAC audio, often used in Chinese digital television and broadcast applications. When targeting AIFF output, FFmpeg fully decodes the AAC audio stream before writing raw PCM data — this is a full transcode of the audio, not a remux, so there is a decode/encode pass involved. The pcm_s16be codec writes samples in big-endian byte order, which is the byte order historically required by the AIFF specification (originally designed for Motorola 68000-based Macs). Metadata from the CAVS container — such as title or creation timestamps — is generally not carried over to AIFF, since AIFF has limited metadata support and the two formats use entirely different tagging schemes. No subtitle, chapter, or secondary audio track data is preserved, as AIFF supports none of these features. If the source CAVS file has a sample rate other than 44.1 kHz (e.g., 48 kHz from broadcast sources), FFmpeg will preserve that sample rate in the AIFF output without resampling unless you explicitly add '-ar 44100' to the command.