Convert WEBA to DSS — Free Online Tool
Convert WEBA audio files (WebM audio containers with Opus or Vorbis encoding) to DSS format using the ADPCM IMA OKI codec — the proprietary compressed audio standard used by Olympus, Philips, and Grundig digital dictation devices. This tool runs entirely in your browser with no file uploads required.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your WEBA file here
or click to browse
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
WEBA files use a WebM container holding high-quality Opus or Vorbis audio, optimized for web streaming and general-purpose listening. DSS (Digital Speech Standard) uses a fundamentally different codec — ADPCM IMA OKI — a low-bitrate adaptive delta pulse-code modulation variant tuned for narrow-band speech. During conversion, FFmpeg decodes the Opus or Vorbis stream from the WEBA container, passes the raw PCM audio through its internal pipeline, and then re-encodes it using the adpcm_ima_oki codec into a DSS container. Because ADPCM IMA OKI targets speech at a fixed low bitrate and narrow frequency range, the output will be significantly smaller but optimized solely for voice intelligibility rather than music or wideband audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg binary, the open-source multimedia processing engine that handles decoding the WEBA container and re-encoding to DSS. This tool runs FFmpeg compiled to WebAssembly (FFmpeg.wasm) entirely inside your browser. |
-i input.weba
|
Specifies the input file — a WEBA audio container, which FFmpeg opens and demuxes to extract the encoded Opus or Vorbis audio stream for decoding. |
-c:a adpcm_ima_oki
|
Instructs FFmpeg to encode the audio output using the ADPCM IMA OKI codec, the fixed low-bitrate adaptive delta PCM codec required by the DSS dictation format. This replaces the high-quality Opus or Vorbis encoding from the source with a narrowband speech codec. |
output.dss
|
Defines the output filename with the .dss extension, which tells FFmpeg to use its DSS muxer to wrap the adpcm_ima_oki encoded audio into a Digital Speech Standard container compatible with Olympus, Philips, and Grundig dictation systems. |
Common Use Cases
- Transferring a browser-recorded voice memo or web conference audio clip into a DSS file for import into dictation transcription software like Olympus DSS Player or Philips SpeechExec.
- Converting a WEBA-format interview recording downloaded from a web platform so it can be played back on a legacy Olympus or Grundig digital dictation recorder.
- Preparing voice notes captured via a WebRTC-based tool for compatibility with legal or medical transcription workflows that mandate DSS format submission.
- Archiving a speech-only WEBA recording into DSS to dramatically reduce file size when long-term storage of intelligible voice content is the priority over audio fidelity.
- Migrating a batch of web-sourced audio dictations from a modern browser-based recording app into DSS so they can be processed by enterprise transcription hardware that only accepts DSS input.
- Testing DSS output from a WEBA source to verify codec compatibility before committing to a large-scale dictation archive conversion pipeline.
Frequently Asked Questions
Yes, and this is expected. WEBA with Opus can carry wideband audio up to 128kbps or higher with excellent perceptual quality, while DSS using ADPCM IMA OKI is a narrowband, low-bitrate codec designed strictly for speech intelligibility on dictation devices. Music, sound effects, or any non-speech content will sound noticeably degraded or tinny. For voice recordings, the result should remain clearly intelligible, which is the format's intended purpose.
The DSS format with the ADPCM IMA OKI codec operates at a fixed encoding configuration — there is no variable bitrate or quality parameter exposed by FFmpeg for this codec. Unlike the WEBA input which supports bitrate options like 64k to 320k, the DSS output's bitrate and quality are determined entirely by the codec's fixed design. This is typical of dictation-specific formats where hardware compatibility requires strict adherence to a single encoding profile.
Not really. The ADPCM IMA OKI codec in DSS is designed around speech frequency ranges, typically emphasizing the 300Hz–3400Hz telephone-quality band. Music with a wide dynamic range, stereo content, or significant high-frequency detail will be heavily compressed and narrowed in this process. If you need to convert music for archival or playback purposes, DSS is not an appropriate target format — it exists specifically for voice dictation.
No. WEBA (WebM) can carry metadata tags such as title, artist, and encoder information, but the DSS container format does not have a standardized general-purpose metadata tagging system equivalent to what WebM supports. FFmpeg will not transfer these tags to the DSS output, so any embedded metadata in the source WEBA file will be lost during conversion.
Copy the FFmpeg command displayed on this page — 'ffmpeg -i input.weba -c:a adpcm_ima_oki output.dss' — and run it in a terminal on any system where FFmpeg is installed. Replace 'input.weba' with the actual path to your file and 'output.dss' with your desired output path. FFmpeg handles files of any size on the desktop without the 1GB browser limitation, and the command is identical to what this tool runs internally.
FFmpeg does not natively batch-process multiple files in a single command, but you can use a shell loop to achieve this. On Linux or macOS, run: 'for f in *.weba; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.weba}.dss"; done'. On Windows PowerShell, use: 'Get-ChildItem *.weba | ForEach-Object { ffmpeg -i $_.Name -c:a adpcm_ima_oki ($_.BaseName + ".dss") }'. This applies the same adpcm_ima_oki encoding to every WEBA file in the directory.
Technical Notes
The DSS format was co-developed by Olympus, Philips, and Grundig and exists in two revisions: DSS (original) and DSS Pro. FFmpeg's DSS muxer with the adpcm_ima_oki codec targets the original DSS standard, which uses a 4-bit ADPCM variant tuned to OKI's telephony codec lineage. The sample rate is fixed at 8000Hz mono, meaning any stereo or high-sample-rate WEBA audio will be downmixed to mono and downsampled during encoding — this is not a limitation of this tool but an inherent constraint of the DSS format itself. WEBA files using the Vorbis codec rather than Opus will also be fully decoded before re-encoding, as there is no codec-copy path possible between these two very different audio formats. Because both WEBA and DSS are lossy formats, this conversion represents a second generation of lossy encoding, which compounds artifacts — particularly at frequencies outside the speech band. Users should avoid converting high-quality WEBA recordings through DSS if there is any possibility of needing to re-encode them again later. DSS files produced by FFmpeg may not expose all proprietary DSS header fields that dedicated Olympus or Philips hardware writes, so compatibility with every piece of dictation software should be tested before committing to a large batch conversion.