Convert CAVS to DSS — Free Online Tool
Convert CAVS video files to DSS audio format, extracting and re-encoding the audio stream using the ADPCM IMA OKI codec optimized for speech and digital dictation devices. This tool strips the Chinese standard video entirely and produces a compact DSS file compatible with Olympus, Philips, and Grundig dictation hardware.
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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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 contain a video stream encoded with H.264 (libx264) and an AAC audio track. During this conversion, the video stream is completely discarded — DSS is a audio-only format with no video container support. The AAC audio is then decoded and re-encoded using the ADPCM IMA OKI codec, a lossy adaptive delta pulse-code modulation scheme specifically designed for low-bitrate speech recording on digital dictation devices. The resulting DSS file is highly compressed and speech-optimized, meaning it will preserve vocal clarity but is not suitable for music or high-fidelity audio. Because DSS has no configurable quality parameters within FFmpeg, the output bitrate and sample rate are fixed by the codec specification.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary — the open-source multimedia processing engine that powers this conversion both in the browser (via FFmpeg.wasm) and on the desktop command line. |
-i input.cavs
|
Specifies the input file in CAVS (Chinese Audio Video Standard) format. FFmpeg demuxes this container to access the H.264 video stream and AAC audio stream separately; only the audio stream proceeds to the output. |
-c:a adpcm_ima_oki
|
Sets the audio codec to ADPCM IMA OKI — the specific adaptive delta PCM variant used in DSS files developed by Olympus, Philips, and Grundig. This re-encodes the decoded AAC audio into the fixed-bitrate, speech-optimized encoding required by the DSS format. |
output.dss
|
Defines the output file with the .dss extension, signaling FFmpeg to write a Digital Speech Standard container. Because DSS is audio-only, the video stream from the CAVS input is automatically dropped — no explicit video discard flag is needed since the DSS muxer accepts no video. |
Common Use Cases
- Extracting spoken commentary or narration from a CAVS broadcast recording to load onto an Olympus or Philips digital dictation recorder for transcription
- Converting CAVS news segment audio into DSS format for archiving spoken-word content in a compact, dictation-device-compatible format
- Preparing interview audio captured in CAVS format for import into digital dictation transcription software that only accepts DSS files
- Stripping a CAVS video lecture or presentation to a DSS audio file for playback on legacy Grundig dictation hardware without video support
- Converting CAVS voice-over recordings to DSS for integration into a professional transcription workflow that uses DSS as its standard interchange format
Frequently Asked Questions
No — this is a lossy-to-lossy conversion, meaning quality is lost at each encoding stage. The AAC audio in the CAVS file must be fully decoded and then re-encoded using the ADPCM IMA OKI codec. DSS is deliberately designed for low-bitrate speech and is not suitable for music or broad-spectrum audio. If your CAVS file contains speech, the result will be intelligible but noticeably lower fidelity than the original.
DSS (Digital Speech Standard) is a proprietary audio-only format developed for digital dictation devices — it has no video track support whatsoever. The conversion intentionally discards the CAVS video stream entirely. If you need to retain video, DSS is the wrong target format; consider formats like MP4 or MKV instead.
No. The DSS format and its ADPCM IMA OKI codec do not expose configurable quality or bitrate parameters in FFmpeg. The output characteristics are fixed by the codec specification, which is optimized for speech intelligibility at a low, fixed bitrate. This is by design — DSS was engineered for consistent behavior across dictation hardware from Olympus, Philips, and Grundig.
Copy the FFmpeg command displayed on this page — 'ffmpeg -i input.cavs -c:a adpcm_ima_oki output.dss' — and run it in a terminal on any machine with FFmpeg installed. This is identical to what the browser tool executes and will handle arbitrarily large CAVS files without any upload or file size restriction. FFmpeg is available free for Windows, macOS, and Linux from ffmpeg.org.
Metadata compatibility between CAVS and DSS is extremely limited. DSS is a proprietary format with a fixed header structure designed for dictation metadata (author ID, work type, priority flags) rather than general media tags. Standard metadata fields from the CAVS container such as title or creation date will not be meaningfully preserved in the DSS output.
FFmpeg itself does not natively batch process files in a single command, but you can easily loop over multiple files using a shell script. On Linux or macOS, run: 'for f in *.cavs; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.cavs}.dss"; done'. On Windows PowerShell, use: 'Get-ChildItem *.cavs | ForEach-Object { ffmpeg -i $_.Name -c:a adpcm_ima_oki ($_.BaseName + ".dss") }'. This applies the same ADPCM IMA OKI encoding to every CAVS file in the directory.
Technical Notes
The ADPCM IMA OKI codec used in DSS files operates at a fixed low sample rate (typically 8000 Hz mono), which is the fundamental reason this format is unsuitable for anything other than speech. CAVS files typically carry AAC audio at 44.1 kHz or 48 kHz stereo, so the conversion involves both downsampling and channel reduction — a significant lossy transformation. The CAVS container itself is a Chinese national broadcast standard that wraps H.264 video and AAC audio; FFmpeg can demux it, but the format is far less common outside China and specialized broadcast environments. Note that DSS has two variants (DSS classic and DSS Pro), and FFmpeg's adpcm_ima_oki implementation targets the classic DSS specification. Compatibility with DSS Pro workflows or newer Olympus firmware may be limited. There is no subtitle, chapter, or secondary audio track data to worry about in this conversion — neither format supports those features.