Convert 3GPP to DSS — Free Online Tool
Convert 3GPP mobile video files to DSS (Digital Speech Standard) audio, extracting and re-encoding the audio stream using the ADPCM IMA OKI codec — the same low-bitrate, speech-optimized encoding used by Olympus and Philips professional dictation devices. Ideal for repurposing recorded mobile conversations or voice memos into a format compatible with digital transcription workflows.
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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 from the 3GPP container entirely and extracts only the audio track. The source audio — typically AAC or MP3 encoded at mobile-optimized bitrates — is fully decoded and then re-encoded using the ADPCM IMA OKI codec, a variant of Adaptive Differential Pulse-Code Modulation developed for Olympus digital dictation hardware. The output is wrapped in a DSS container, a proprietary format jointly standardized by Olympus, Philips, and Grundig. Because DSS is purpose-built for speech at very low bitrates, the codec applies aggressive compression optimized for vocal frequency ranges, meaning music or complex audio in the source file will sound degraded but speech content will remain intelligible.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. In the browser-based version of this tool, FFmpeg runs locally via WebAssembly (FFmpeg.wasm) — no data leaves your device. |
-i input.3gp
|
Specifies the input file in 3GPP format — a mobile multimedia container that typically holds H.264 video and AAC audio. FFmpeg reads both streams but only the audio will be used in this conversion. |
-c:a adpcm_ima_oki
|
Instructs FFmpeg to encode the output audio using the ADPCM IMA OKI codec — the proprietary speech codec required by the DSS format and used in Olympus and Philips digital dictation hardware. The source audio (typically AAC from the 3GPP file) is fully decoded and re-encoded at 8000 Hz using this codec. |
output.dss
|
Defines the output filename and container. The .dss extension tells FFmpeg to wrap the ADPCM IMA OKI audio stream in a Digital Speech Standard container, discarding the video stream from the 3GPP source entirely since DSS is an audio-only format. |
Common Use Cases
- Sending a mobile phone voice recording to a legal or medical transcription service that requires DSS files compatible with Olympus or Philips dictation software
- Archiving field interviews recorded on a smartphone in 3GPP format into DSS for ingestion into a digital transcription management system
- Converting a 3GPP video of a deposition or oral statement into a DSS audio file for use with speech-recognition dictation software
- Extracting the audio from a 3GPP meeting recording to produce a compact, speech-optimized DSS file for offline review on a dedicated digital dictaphone player
- Migrating legacy mobile recordings captured on older 3G-era handsets into DSS format required by enterprise document workflow systems
- Preparing voice memos from a mobile device for import into transcription platforms like Olympus DSS Player or Philips SpeechExec
Frequently Asked Questions
Yes, there will be a quality reduction. DSS using the ADPCM IMA OKI codec is engineered specifically for speech intelligibility at very low bitrates — it is not designed to preserve music, stereo imaging, or broad frequency ranges. If your 3GPP file contains speech, the result will typically remain clear and usable for transcription purposes. However, if the source contains music, background noise, or complex audio, those elements will sound noticeably degraded. Both formats are lossy, so this conversion involves a lossy-to-lossy transcode with no opportunity to recover detail lost in the original 3GPP encoding.
No. The DSS format has extremely limited metadata support compared to modern containers, and the ADPCM IMA OKI codec carries no provisions for embedding tags like title, date, or artist. Any metadata present in the 3GPP container — such as creation timestamps or device information — will not be carried over to the DSS output. If metadata preservation is important, you should document it separately before converting.
DSS files using the ADPCM IMA OKI codec operate at a fixed sample rate of 8000 Hz, which is narrowband telephone quality. If your 3GPP source audio was recorded at a higher sample rate — commonly 44100 Hz or 48000 Hz on modern phones — FFmpeg will automatically downsample the audio to 8 kHz during the conversion. This downsampling is a significant reduction and removes high-frequency content above 4 kHz, which is why the format is only appropriate for speech rather than music or wideband audio.
No — DSS with the ADPCM IMA OKI codec does not support variable bitrate or quality parameters. The codec encodes at a fixed rate dictated by the DSS specification, so flags like -b:a (bitrate) or -q:a (quality) have no effect and should not be added to the command. The command as shown — ffmpeg -i input.3gp -c:a adpcm_ima_oki output.dss — is essentially the complete and only practical form for this conversion.
FFmpeg does not natively batch process multiple files in a single command, but you can script it easily. On Linux or macOS, use a shell loop: for f in *.3gp; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.3gp}.dss"; done. On Windows Command Prompt, use: for %f in (*.3gp) do ffmpeg -i "%f" -c:a adpcm_ima_oki "%~nf.dss". This is especially useful for converting large archives of mobile recordings to DSS in one operation.
DSS is a closed, hardware-centric proprietary format developed in the 1990s for dedicated digital dictation recorders, long before AAC became standard. Its only supported codec is ADPCM IMA OKI, a simple adaptive PCM variant chosen for low computational overhead on embedded hardware. There is no mechanism in the DSS specification to carry AAC, MP3, or any modern codec — so full decoding and re-encoding to ADPCM IMA OKI is unavoidable when converting from 3GPP.
Technical Notes
The ADPCM IMA OKI codec used in DSS is a variant of IMA ADPCM adapted by Olympus for their dictation device lineup, encoding audio at a fixed 8000 Hz sample rate with 4 bits per sample, yielding approximately 32 kbps. This is substantially narrower than even the most conservative AAC encoding found in 3GPP files. FFmpeg's DSS muxer and ADPCM IMA OKI encoder have limited adoption compared to mainstream formats, so compatibility should be verified with the target dictation software (e.g., Olympus DSS Player Pro, Philips SpeechExec) before deploying in a production workflow. Notably, DSS has two sub-variants (DSS and DSS Pro/DS2); this tool produces standard DSS, not the newer DS2 format. The 3GPP video track is silently dropped with no special flag required, since DSS is an audio-only container. If the 3GPP source has multiple audio tracks, FFmpeg will select the default (first) audio track automatically.