Convert CAVS to WAV — Free Online Tool
Extract and convert audio from CAVS video files into uncompressed WAV format using PCM 16-bit signed little-endian encoding. This tool strips the AAC audio track from Chinese Audio Video Standard files and outputs a lossless-quality WAV file — ideal for archiving or professional audio workflows that require uncompressed PCM data.
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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 typically carry AAC-encoded audio alongside an H.264 video stream. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio track back to raw PCM samples, which are then written into a WAV container using the pcm_s16le codec — 16-bit signed integers in little-endian byte order, sampled at whatever rate the source audio uses. This is a full decode-then-re-encode operation on the audio: AAC is a lossy format, so the original lossy compression is decoded and the resulting PCM data is stored without further compression in WAV. No additional quality is lost beyond what AAC already removed during the original CAVS encoding. The output WAV file will be significantly larger than the CAVS source because uncompressed PCM audio takes far more space than AAC.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles all decoding, stream selection, and encoding in this conversion. |
-i input.cavs
|
Specifies the input file in CAVS format. FFmpeg reads the container, identifies the H.264 video stream and the AAC audio stream inside it, and makes both available for processing. |
-c:a pcm_s16le
|
Instructs FFmpeg to encode the audio output using PCM signed 16-bit little-endian — the standard uncompressed audio codec for WAV files. This decodes the AAC audio from the CAVS source into raw PCM samples without any further compression. |
output.wav
|
Defines the output filename and container format. The .wav extension tells FFmpeg to use the WAV container, which wraps the pcm_s16le audio data in a RIFF/WAV structure compatible with virtually all audio software and operating systems. |
Common Use Cases
- Extracting dialogue or commentary audio from CAVS broadcast recordings for use in a professional audio editor like Audacity, Adobe Audition, or DaVinci Resolve Fairlight, which require uncompressed PCM input
- Archiving the audio component of Chinese broadcast or IPTV CAVS recordings in a universally readable format before the source files are deleted or reformatted
- Feeding CAVS-sourced audio into speech recognition or transcription pipelines that require raw PCM WAV input rather than compressed AAC streams
- Preparing audio from CAVS video files for quality analysis or forensic audio examination, where uncompressed WAV is necessary to avoid introducing additional compression artifacts
- Converting CAVS event or conference recordings to WAV so the audio can be mastered, noise-reduced, or re-encoded to MP3/FLAC for podcast or distribution purposes
- Extracting audio from CAVS files for use in video game engines or interactive media tools that accept WAV but do not support AAC or the CAVS container
Frequently Asked Questions
Not in the strict sense. AAC is a lossy codec, meaning some audio information was permanently discarded when the CAVS file was originally encoded. This conversion decodes the AAC stream and writes it to PCM WAV without any additional quality loss — but it cannot recover what AAC already threw away. The resulting WAV is an exact, uncompressed representation of the decoded AAC audio, which is as good as the source allows.
CAVS files store audio as AAC, which is highly compressed — typically at 128 kbps. WAV using pcm_s16le stores every audio sample as a raw 16-bit integer with no compression at all. For stereo audio at 44.1 kHz, that works out to roughly 10 MB per minute, compared to around 1 MB per minute for 128 kbps AAC. A 5-minute CAVS clip with 5 MB of audio could produce a 50 MB WAV file.
No. WAV is a pure audio container and cannot hold video data. FFmpeg automatically drops the H.264 video stream from the CAVS file and extracts only the audio track. If you need to keep the video, you should choose a video output format like MP4 or MKV instead.
Yes. The default command uses pcm_s16le (16-bit signed little-endian), which is standard for most applications. To get 24-bit output, modify the FFmpeg command to use -c:a pcm_s24le instead. For 32-bit float output, use -c:a pcm_f32le. Keep in mind that the original AAC audio in the CAVS file was likely encoded at 16-bit effective depth anyway, so going higher than 16-bit adds container headroom but not real audio precision.
On Linux or macOS, you can use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -c:a pcm_s16le "${f%.cavs}.wav"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -c:a pcm_s16le "%~nf.wav". This processes each CAVS file in the current directory and creates a matching WAV file with the same base filename.
Yes. By default, FFmpeg preserves the original sample rate and channel layout from the AAC audio in the CAVS file. If the source was stereo 48 kHz (common in broadcast CAVS), the WAV output will also be stereo 48 kHz. You can override this with -ar (e.g., -ar 44100 for 44.1 kHz) or -ac (e.g., -ac 1 for mono) if your downstream workflow requires a specific format.
Technical Notes
CAVS (GB/T 20090) is a Chinese national standard primarily used in broadcast and IPTV contexts within China. Its audio is almost universally AAC, encoded in the same way as in MP4 or TS containers, making audio extraction straightforward for FFmpeg. The pcm_s16le codec chosen for the WAV output is the most universally compatible PCM format — it is the default expected by Windows audio APIs, most DAWs, and legacy hardware samplers. One notable limitation of this conversion is that any metadata embedded in the CAVS file (such as title or language tags) is unlikely to be preserved, as WAV has very limited metadata support and CAVS metadata structures are not standard. If the CAVS source contains multiple audio tracks (unusual but possible in broadcast recordings), only the first audio track will be extracted by default; use -map 0:a:1 to select an alternate track. Chapter and subtitle data are irrelevant here since both CAVS and WAV lack meaningful support for those features in this context.