Extract Audio from CAVS to WAV — Free Online Tool
Extract audio from CAVS (Chinese Audio Video Standard) video files and save it as uncompressed WAV audio. This tool strips the AAC audio track from your CAVS file and re-encodes it to 16-bit PCM (pcm_s16le), giving you a lossless-quality WAV file ready for professional audio editing or archiving.
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 an AAC-encoded audio track alongside H.264 video. During this conversion, FFmpeg discards the video stream entirely using the -vn flag, then decodes the compressed AAC audio from its lossy compressed state and re-encodes it into 16-bit signed little-endian PCM — the standard uncompressed audio format used inside WAV containers. Because AAC is a lossy codec, the original audio has already undergone one round of compression; the WAV output preserves that decoded audio at full fidelity without any further quality loss, giving you the best possible extraction from what the CAVS file contains.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, which handles all demuxing, decoding, and encoding operations in this conversion. In the browser version of this tool, FFmpeg runs as a WebAssembly module (FFmpeg.wasm) locally on your device — no data is sent to a server. |
-i input.cavs
|
Specifies the input file — a CAVS-formatted video file containing an H.264 video stream and an AAC audio track. FFmpeg uses the cavs demuxer to read and parse this container format. |
-vn
|
Disables video output entirely, telling FFmpeg to ignore the H.264 video stream from the CAVS file. This is essential because WAV is an audio-only container and cannot carry video data. |
-c:a pcm_s16le
|
Decodes the AAC audio from the CAVS file and re-encodes it as 16-bit signed little-endian PCM — the standard uncompressed audio codec used in WAV files. This produces the most widely compatible WAV output, readable by virtually all audio software and devices. |
output.wav
|
Defines the output filename and tells FFmpeg to write the result as a WAV container (RIFF/WAV format). The .wav extension causes FFmpeg to use the WAV muxer, which wraps the pcm_s16le audio stream in the standard WAV file structure. |
Common Use Cases
- Extracting a dialogue or commentary track from a CAVS broadcast recording for use in subtitle or dubbing workflows that require uncompressed WAV input
- Pulling audio from Chinese digital television or CAVS-encoded content to import into a DAW like Audacity or Adobe Audition, which work best with uncompressed PCM audio
- Archiving the audio component of CAVS broadcast footage separately from the video, since WAV is a universally readable, long-term storage format
- Preparing audio from CAVS files for transcription or speech recognition software that requires raw, uncompressed WAV input for accurate processing
- Isolating a musical score or sound effects track from a CAVS-encoded production for remixing or sound design work
- Converting CAVS audio to WAV as an intermediate step before further encoding to formats like MP3 or FLAC, avoiding additional generation loss by starting from the decoded PCM signal
Frequently Asked Questions
No — the audio quality is capped by the original AAC encoding inside the CAVS file. AAC is a lossy codec, so some audio detail was permanently discarded when the CAVS file was created. Converting to WAV decodes that AAC audio to uncompressed PCM, which means no further quality is lost in this step, but the WAV output will not be higher quality than the original AAC source. WAV is ideal here because it ensures no additional compression artifacts are introduced.
CAVS files store audio as compressed AAC, which typically achieves 10:1 or greater compression ratios compared to uncompressed audio. When FFmpeg decodes that AAC track and writes it to WAV as 16-bit PCM, the audio is stored at its full, uncompressed size — roughly 10 MB per minute for stereo audio at 44.1kHz. The CAVS file also included a compressed H.264 video stream, so the resulting WAV will only represent the audio portion, but that audio portion alone will be significantly larger than what AAC occupied in the original container.
CAVS files are a Chinese broadcast standard and typically carry minimal embedded metadata compared to consumer formats. WAV's metadata support is also limited — it uses INFO chunks or ID3 tags, but FFmpeg's CAVS demuxer may not map broadcast metadata fields into WAV-compatible tags. You should not rely on this conversion to transfer metadata; use a dedicated tag editor like Mp3tag after conversion if metadata fields are important to your workflow.
To resample the audio, add -ar followed by the target sample rate (e.g., -ar 44100 or -ar 48000) before the output filename. To change the bit depth, swap pcm_s16le for another codec: pcm_s24le gives you 24-bit audio, pcm_s32le gives 32-bit, and pcm_u8 gives 8-bit. For example: ffmpeg -i input.cavs -vn -c:a pcm_s24le -ar 48000 output.wav. Note that upsampling beyond the original AAC encoding rate does not add real audio information.
Yes. On Linux or macOS, you can use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a pcm_s16le "${f%.cavs}.wav"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a pcm_s16le "%~nf.wav". This processes each CAVS file in the directory and outputs a matching WAV file, which is especially useful when working with a series of broadcast segments.
pcm_s16le stands for 16-bit signed integer PCM in little-endian byte order, which is the most universally compatible WAV format — it is what standard audio CDs use and what virtually every application and device expects when opening a WAV file. If you are delivering to professional audio or broadcast workflows that require 24-bit resolution, substitute pcm_s24le. If you are working with consumer tools or need the smallest possible uncompressed file, pcm_s16le is the correct default choice.
Technical Notes
CAVS (Chinese Audio Video Standard, GB/T 20090) is a broadcast-oriented container used primarily in Chinese digital television and media distribution. Its audio track is encoded as AAC, most commonly at 128kbps stereo at 48kHz — the broadcast standard sample rate. When FFmpeg demuxes the CAVS container and decodes the AAC stream to PCM, the output sample rate matches whatever the AAC track was encoded at (typically 48kHz), so your WAV file will reflect that rate unless you explicitly resample. The -vn flag is critical here because without it, FFmpeg would attempt to handle the H.264 video stream, which is not mapped into a WAV container and would cause an error. CAVS support in FFmpeg relies on the cavs demuxer, which is well-established but may have limited handling of non-standard or proprietary CAVS variants from certain Chinese broadcast equipment. If you encounter decoding errors, check that your FFmpeg build includes the cavs demuxer by running ffmpeg -formats | grep cavs. The resulting WAV file uses a standard RIFF/WAV structure and will open in any audio editor, DAW, or transcription tool without compatibility issues.