Extract Audio from 3G2 to DSS — Free Online Tool
Extract audio from 3G2 mobile video files and convert it to DSS (Digital Speech Standard) format using the ADPCM IMA OKI codec — the same codec used in Olympus, Philips, and Grundig digital dictation devices. This tool is ideal for repurposing voice recordings or spoken-word content captured on older CDMA mobile devices into a format compatible with professional dictation workflows.
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FFmpeg Command
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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 strips the video stream entirely from the 3G2 container and re-encodes the audio — which is typically AAC in a 3G2 file — into ADPCM IMA OKI, the proprietary codec at the heart of the DSS format. This is a full audio transcode, not a remux: the AAC audio data must be decoded to raw PCM first, then re-encoded using the OKI ADPCM algorithm, which applies adaptive delta pulse-code modulation optimized for low-bitrate speech. The resulting DSS file is a compact, speech-optimized mono audio file suited for digital dictation playback devices and transcription software.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg application, the open-source multimedia processing engine that handles decoding the 3G2 container, stripping the video, transcoding the AAC audio, and writing the DSS output file. |
-i input.3g2
|
Specifies the input file — a 3G2 container typically holding H.264 video and AAC audio recorded on a CDMA mobile device. FFmpeg will demux this container to access its audio stream for transcoding. |
-vn
|
Disables video output entirely, which is required here because DSS is a pure audio format and cannot contain a video stream. Without this flag, FFmpeg would attempt to include video in the output and fail, since DSS has no video codec support. |
-c:a adpcm_ima_oki
|
Instructs FFmpeg to encode the audio using the ADPCM IMA OKI codec — the specific ADPCM variant used by the DSS format for Olympus, Philips, and Grundig digital dictation devices. This triggers a full transcode from the source AAC audio to the OKI waveform codec. |
output.dss
|
Specifies the output filename with the .dss extension, telling FFmpeg to write the result as a Digital Speech Standard file. FFmpeg infers the DSS container format from this extension and packages the ADPCM IMA OKI audio stream accordingly. |
Common Use Cases
- Transferring voice memos or spoken-word recordings captured on a CDMA mobile phone in 3G2 format into a dictation system that accepts DSS files, such as Olympus or Philips transcription software.
- Archiving field interviews or verbal notes recorded on older 3G2-capable handsets into DSS for long-term storage in a dictation-compatible archive.
- Converting 3G2 audio content from legacy mobile recordings to DSS so it can be loaded onto a standalone digital dictation device for playback without a smartphone.
- Extracting the audio track from a 3G2 video captured during a meeting or lecture and converting it to DSS for import into a transcription workflow that only accepts dictation formats.
- Repurposing 3G2 voice recordings from CDMA network-era phones into DSS format for use with foot-pedal-controlled transcription software used in legal or medical settings.
Frequently Asked Questions
Yes, there will be quality loss. The audio in a 3G2 file is typically encoded as AAC at 128kbps or higher, which is a modern lossy codec with reasonably good fidelity. Converting to DSS uses the ADPCM IMA OKI codec, which is a much older, lower-fidelity codec designed purely for compressing speech at very low bitrates. Music or complex audio will sound noticeably degraded in DSS, but spoken voice content — which is what DSS was designed for — will remain intelligible and usable for transcription purposes.
They are not directly compatible, which is why FFmpeg must fully transcode the audio rather than simply copying the stream. AAC is a modern transform-based codec operating in the frequency domain, while ADPCM IMA OKI is a time-domain waveform codec based on adaptive delta modulation. FFmpeg decodes the AAC audio to raw PCM, then re-encodes it using the OKI ADPCM algorithm to produce the DSS file. This two-step decode-encode process is handled automatically by the command.
No. The DSS format with the ADPCM IMA OKI codec does not support configurable bitrate or quality parameters through FFmpeg — the codec operates at a fixed encoding specification. This is by design, as DSS was built for dictation devices with fixed hardware decoders. The output quality is determined entirely by the characteristics of the OKI ADPCM codec itself, not by any user-adjustable setting. If the audio quality from your 3G2 source is poor, it cannot be improved in the DSS output.
The ADPCM IMA OKI codec used in DSS files typically operates at 8000 Hz (8kHz), which is standard telephone-quality audio. If your 3G2 file contains audio recorded at a higher sample rate (e.g., 44.1kHz or 48kHz), FFmpeg will automatically downsample it to match the DSS codec's requirements during conversion. This downsampling is another reason why music sounds poor in DSS, but it has minimal impact on speech intelligibility.
You can batch process files on the command line using a shell loop. On Linux or macOS, run: `for f in *.3g2; do ffmpeg -i "$f" -vn -c:a adpcm_ima_oki "${f%.3g2}.dss"; done`. On Windows Command Prompt, use: `for %f in (*.3g2) do ffmpeg -i "%f" -vn -c:a adpcm_ima_oki "%~nf.dss"`. The browser-based tool on this page processes one file at a time, so the FFmpeg command is the recommended approach for bulk conversions.
Metadata preservation is very limited in this conversion. DSS is a proprietary format developed for dictation hardware and has a minimal, device-specific metadata structure that does not map cleanly to the metadata tags used in 3G2 containers. FFmpeg will not transfer tags like artist, title, or creation date from the 3G2 file into the DSS output. If metadata retention is important to your workflow, consider maintaining the original 3G2 file as the archival source alongside the converted DSS file.
Technical Notes
The DSS (Digital Speech Standard) format is a proprietary specification jointly developed by Olympus, Philips, and Grundig, and it is tightly coupled to the ADPCM IMA OKI codec — there is no alternative audio codec available for DSS output in FFmpeg. The OKI ADPCM variant used here encodes audio at a fixed low bitrate with a sample rate of 8000 Hz and mono channel configuration, which means any stereo audio in the source 3G2 file will be downmixed to mono. The 3G2 container format, originally designed for CDMA networks (3GPP2 standard), stores audio most commonly as AAC, so this conversion always involves a full transcode from AAC to ADPCM IMA OKI — there is no possibility of stream copying. Because both formats are lossy, this is a lossy-to-lossy transcode, meaning generational quality loss is unavoidable. DSS files produced by FFmpeg may not be 100% identical in header structure to files produced by native Olympus or Philips dictation devices, and compatibility with all dictation software or hardware should be tested before use in a production transcription workflow. The -vn flag is essential here because DSS is an audio-only format and cannot carry a video stream.