Extract Audio from CAVS to DSS — Free Online Tool
Extract audio from CAVS (Chinese Audio Video Standard) video files and convert it to DSS (Digital Speech Standard) format using the ADPCM IMA OKI codec — the native encoding used by Olympus, Philips, and Grundig digital dictation devices. This tool is particularly useful for repurposing broadcast or archival CAVS content into a speech-optimized, low-bitrate format compatible with professional dictation workflows.
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FFmpeg Command
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Estimated output:
Conversion Complete!
DownloadHow It Works
CAVS files typically carry AAC-encoded audio alongside H.264 video. During this conversion, the video stream is completely discarded and never processed — FFmpeg reads only the audio track. That audio is then decoded from AAC and re-encoded using the ADPCM IMA OKI codec, a lossy adaptive delta pulse-code modulation algorithm specifically designed for intelligible speech reproduction at very low bitrates. DSS is a fixed-spec format developed jointly by Olympus, Philips, and Grundig, so there are no user-selectable bitrate or quality options — the output parameters are dictated entirely by the DSS container specification. The result is a compact, speech-optimized audio file ready for use with digital dictation software or hardware.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In this browser-based tool, the same binary runs locally in your browser via WebAssembly (FFmpeg.wasm), so no files leave your device. |
-i input.cavs
|
Specifies the input file in CAVS format. FFmpeg reads the container to locate both the H.264 video and AAC audio streams, though only the audio stream will be used in this conversion. |
-vn
|
Disables video output entirely. This tells FFmpeg to ignore the H.264 video track from the CAVS file and produce an audio-only output, which is required since DSS is a pure audio format with no video support. |
-c:a adpcm_ima_oki
|
Encodes the audio using the ADPCM IMA OKI codec, the native encoding algorithm of the DSS (Digital Speech Standard) format used by Olympus, Philips, and Grundig dictation devices. This re-encodes the source AAC audio from the CAVS file at a fixed 8000 Hz mono rate optimized for speech intelligibility. |
output.dss
|
Sets the output filename with the .dss extension, which tells FFmpeg to mux the encoded ADPCM IMA OKI audio into a DSS container — the proprietary dictation file format expected by digital dictation playback software and hardware. |
Common Use Cases
- Extracting spoken commentary or narration from a CAVS broadcast recording to load into a professional dictation transcription system like Olympus DSS Player
- Repurposing CAVS-encoded interview or conference recordings from Chinese broadcast archives into DSS format for playback on Philips SpeechMike devices
- Converting CAVS news segment audio into DSS for ingestion into legacy transcription workflows that only accept dictation-standard file formats
- Archiving the spoken-word audio track from CAVS government or institutional broadcast content in a compact, speech-tuned format for long-term dictation record-keeping
- Preparing CAVS source material for transcription services or legal documentation workflows that require DSS-format audio delivered from digital dictation equipment
- Testing compatibility of CAVS-sourced audio content with Grundig Digta dictation hardware by converting to the native DSS format those devices expect
Frequently Asked Questions
DSS with ADPCM IMA OKI is engineered specifically for speech intelligibility at low bitrates, not for music or high-fidelity audio reproduction. If the CAVS source contains clear spoken dialogue, the output will be intelligible but noticeably compressed compared to the original AAC audio. For music or complex audio content embedded in a CAVS file, expect significant quality loss — DSS is the wrong target format for anything other than voice recordings.
The DSS format is a closed, proprietary specification co-developed by Olympus, Philips, and Grundig. Unlike open formats such as MP3 or AAC, DSS defines fixed encoding parameters as part of the standard itself — there is no mechanism to select a bitrate or quality level. FFmpeg's ADPCM IMA OKI encoder writes the audio according to those fixed parameters, so the output spec is determined by the format, not by user settings.
The DSS format has very limited metadata support by design — it was built for dictation devices, not general media players. Most metadata embedded in the CAVS container, including title tags or language identifiers, will not carry over to the DSS output. If metadata preservation is important, you should store it separately before conversion.
Yes. On Linux or macOS you can wrap the command in a shell loop: `for f in *.cavs; do ffmpeg -i "$f" -vn -c:a adpcm_ima_oki "${f%.cavs}.dss"; done`. On Windows Command Prompt, use `for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a adpcm_ima_oki "%~nf.dss"`. Each file is processed independently, and since DSS has no tunable parameters, the command is identical for every file in the batch.
The ADPCM IMA OKI codec used in DSS files operates at 8000 Hz mono, which is the standard for digital dictation. If the audio track in your CAVS file is at a higher sample rate (44.1 kHz or 48 kHz is common in broadcast content), FFmpeg will automatically resample and downmix it to 8 kHz mono during the conversion. This downsampling is a key reason why DSS is suitable only for speech — musical content or stereo audio will sound degraded.
DSS files produced by FFmpeg using the adpcm_ima_oki codec should be compatible with standard DSS playback software such as Olympus DSS Player, though compatibility can vary between software versions and DSS sub-variants (DSS Classic vs. DSS Pro). If playback fails, the issue is typically that the software expects a specific DSS sub-format header that FFmpeg's muxer may not fully replicate. Testing with your target software before large-scale conversion is recommended.
Technical Notes
CAVS (Chinese Audio Video Standard) files contain H.264 video and AAC audio, both of which are well-supported by FFmpeg. The audio extraction step is straightforward since AAC is a widely decoded format. The complexity in this conversion lies entirely on the output side: DSS is a niche, proprietary container designed for dictation devices, and FFmpeg's support for writing it is functional but limited. The ADPCM IMA OKI codec enforces 8000 Hz mono output, meaning any stereo or high-sample-rate AAC audio from the CAVS source will be forcibly resampled and downmixed — this is not a limitation of FFmpeg's implementation but a hard constraint of the DSS specification. There are no audio quality flags available for the output because the DSS format provides no quality tuning mechanism. File sizes will be very small relative to the source CAVS file, reflecting both the removal of the video stream and the extremely low bitrate of ADPCM IMA OKI encoding. Users should also be aware that DSS has two sub-variants (Classic DSS and DSS Pro), and FFmpeg does not distinguish between them at the muxer level — interoperability with specific dictation hardware or software should be verified before relying on this conversion in a production workflow.