Convert 3G2 to DSS — Free Online Tool
Convert 3G2 mobile video files to DSS (Digital Speech Standard) audio, extracting and re-encoding the audio stream using the ADPCM IMA OKI codec optimized for speech dictation devices from Olympus, Philips, and Grundig. This is useful when you need to bring recorded voice content from a CDMA-era mobile phone into a professional dictation workflow.
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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
During this conversion, FFmpeg discards the video stream entirely from the 3G2 container and extracts only the audio track — which is typically AAC or MP3 encoded at mobile-optimized bitrates. That audio is then transcoded to the ADPCM IMA OKI codec, a low-complexity adaptive delta PCM variant specifically designed for the DSS format used by digital dictation hardware. The DSS format has no configurable audio quality parameters in FFmpeg; the codec operates at a fixed specification tuned for intelligible speech reproduction at very low bitrates. Because both 3G2 and DSS are lossy formats, this is a lossy-to-lossy transcode, meaning generational quality loss will occur — the original AAC audio is decoded to PCM and then re-encoded to ADPCM IMA OKI.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, the open-source multimedia processing engine that handles all decoding, transcoding, and muxing operations in this conversion. In the browser-based tool, this runs via FFmpeg.wasm compiled to WebAssembly. |
-i input.3g2
|
Specifies the input file in 3G2 format — a CDMA mobile multimedia container typically holding H.264 video and AAC audio. FFmpeg will demux this container and make its streams available for processing. |
-c:a adpcm_ima_oki
|
Instructs FFmpeg to encode the audio stream using the ADPCM IMA OKI codec, which is the sole audio codec supported by the DSS format and is specifically designed for low-bitrate speech reproduction on Olympus, Philips, and Grundig digital dictation devices. |
output.dss
|
Defines the output file with the .dss extension, which causes FFmpeg to use the DSS muxer. The video stream from the 3G2 input is implicitly dropped since the DSS format is audio-only and has no video stream support. |
Common Use Cases
- Importing a voice memo or verbal note recorded on an older CDMA mobile phone into an Olympus or Philips dictation management system for transcription by a legal or medical secretary.
- Migrating archived 3G2 voice recordings from a corporate CDMA handset fleet into a DSS-based dictation archive for compliance or records management purposes.
- Preparing a field interview recorded on a 3G2-capable device for playback on a standalone DSS digital dictation machine in an office that does not support modern audio formats.
- Extracting spoken instructions or audio notes captured as 3G2 clips and converting them to DSS so they can be loaded onto an Olympus DS-series recorder for review during commutes.
- Converting 3G2 audio content from legacy mobile surveillance or field reporting devices into DSS format for ingestion into dictation transcription software that only accepts DSS input files.
Frequently Asked Questions
That depends on the content. The ADPCM IMA OKI codec used in DSS is specifically optimized for speech intelligibility at low bitrates, so voice recordings from 3G2 files — such as voice memos or phone calls — generally remain understandable. However, music or complex audio in a 3G2 file will sound noticeably degraded because DSS was never designed for wideband or musical content. Since both 3G2's AAC audio and DSS's ADPCM IMA OKI are lossy codecs, you are performing a lossy-to-lossy transcode with cumulative quality loss.
The DSS format's ADPCM IMA OKI codec operates at a fixed encoding specification defined by the Digital Speech Standard itself — it was designed for dictation hardware with predetermined playback characteristics, not for variable-quality encoding. FFmpeg does not expose bitrate or quality flags for this codec in the DSS container, so the output is always encoded at the codec's native fixed rate. This is by design and consistent with how DSS devices and software expect to receive files.
The video stream is completely dropped during this conversion. DSS is a pure audio format with no support for video, chapters, subtitles, or multiple audio tracks, so FFmpeg extracts only the first audio track from the 3G2 container and discards everything else. If your 3G2 file contains multiple audio tracks, only the default track will be included in the DSS output.
No. If the 3G2 file contains only a video stream and no audio track, FFmpeg will produce an error because there is no audio data to encode into the DSS file. DSS is exclusively an audio format, so a valid audio stream must exist in the source 3G2 file for the conversion to succeed.
On Linux or macOS, you can run a shell loop in your terminal: for f in *.3g2; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.3g2}.dss"; done. On Windows Command Prompt, use: for %f in (*.3g2) do ffmpeg -i "%f" -c:a adpcm_ima_oki "%~nf.dss". This processes each 3G2 file in the current directory and outputs a correspondingly named DSS file. The in-browser tool handles one file at a time, so the FFmpeg command is especially valuable for bulk conversions.
Metadata preservation is very limited in this conversion. DSS is a proprietary dictation format with a fixed header structure designed for device-specific fields like author ID, recorder type, and work type codes — not general-purpose metadata tags like title or artist. Standard metadata embedded in the 3G2 container (such as creation date or device information) will not carry over meaningfully to the DSS file, as FFmpeg's DSS muxer does not map these fields from the source container.
Technical Notes
The 3G2 container is a variant of the MPEG-4 Part 12 file format developed for 3GPP2 CDMA networks, and typically carries H.264 video alongside AAC audio at bitrates constrained by mobile network conditions. DSS (Digital Speech Standard) was jointly developed by Olympus, Philips, and Grundig as a proprietary format for portable dictation recorders, and it uses the ADPCM IMA OKI codec — a variant of Dialogic/OKI ADPCM — which encodes audio at a fixed low bitrate suited for narrowband speech. Because DSS targets speech frequencies, its effective bandwidth is significantly narrower than the AAC audio in a typical 3G2 file; any high-frequency audio content will be lost. The DSS format does not support subtitles, chapters, multiple audio tracks, or transparency, and the FFmpeg DSS muxer has no exposed quality or bitrate parameters. Files produced by this conversion may not be compatible with all DSS playback software depending on which DSS subformat variant is expected by a given dictation system. Testing output files with your specific Olympus, Philips, or Grundig software or device before batch processing is strongly recommended.