Convert FLV to DSS — Free Online Tool
Convert FLV video files to DSS audio format using the ADPCM IMA OKI codec — the same low-bitrate, speech-optimized encoding used in Olympus and Philips digital dictation devices. This tool strips the video stream entirely and extracts only the audio, re-encoding it for compatibility with professional dictation software and transcription workflows.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your FLV 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
FLV files typically contain H.264 video with AAC or MP3 audio. During this conversion, FFmpeg discards the video stream completely and re-encodes only the audio track using the ADPCM IMA OKI codec, which is the proprietary compressed audio format at the core of the DSS (Digital Speech Standard) container. ADPCM IMA OKI is a lossy adaptive delta PCM algorithm tuned for speech intelligibility at very low bitrates — it is not a general-purpose audio codec. The result is a DSS file that can be opened by dictation and transcription applications such as Olympus DSS Player or Philips SpeechExec, but will sound significantly degraded compared to music or wideband audio due to the codec's narrow speech-frequency focus.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg program, which handles all the demuxing, decoding, re-encoding, and muxing required to transform the FLV input into a DSS output file. |
-i input.flv
|
Specifies the input Flash Video file. FFmpeg will demux both the video and audio streams from the FLV container, though only the audio stream will be used in this conversion. |
-c:a adpcm_ima_oki
|
Instructs FFmpeg to encode the audio stream using the ADPCM IMA OKI codec — the specific adaptive delta PCM algorithm required by the DSS (Digital Speech Standard) container format used in Olympus and Philips dictation devices. |
output.dss
|
Defines the output filename with the .dss extension, which tells FFmpeg to mux the encoded audio into a DSS container. The video stream from the FLV is implicitly dropped because the DSS format supports audio only. |
Common Use Cases
- Extract spoken audio from an FLV lecture recording or webinar and convert it to DSS format for import into a professional transcription service that requires Olympus or Philips dictation file formats.
- Transfer a voice memo or interview that was originally captured and shared as a Flash Video file into a DSS file compatible with foot-pedal-controlled transcription software used in legal or medical offices.
- Archive spoken-word FLV recordings — such as old podcast episodes hosted on Flash-era platforms — in DSS format for long-term storage in a dictation management system.
- Prepare an FLV-encoded voiceover or narration clip for review using Olympus DSS Player, where the dictation review workflow requires DSS-formatted files.
- Convert FLV audio commentary from legacy Flash-based e-learning modules into DSS files so transcriptionists can process the spoken content using dictation playback hardware.
Frequently Asked Questions
Expect a significant reduction in audio quality, particularly for music or wideband audio. The DSS format uses the ADPCM IMA OKI codec, which was specifically engineered for speech intelligibility at very low bitrates — not for general audio fidelity. If your FLV source contains clear speech, it will remain intelligible in the DSS output, but the narrow frequency range and heavy compression of ADPCM IMA OKI will make the result sound noticeably thin and lo-fi compared to the original AAC or MP3 audio in the FLV.
The video stream is completely discarded. DSS is a purely audio format with no support for video, so FFmpeg extracts only the audio track from the FLV and re-encodes it as ADPCM IMA OKI. The output DSS file contains no video data whatsoever — it is audio-only by design.
In most cases, yes — DSS files produced with the ADPCM IMA OKI codec are the standard format that Olympus DSS Player and Philips SpeechExec are built to handle. However, some versions of these applications enforce strict header validation or expect files to originate from specific recorder hardware, which may cause compatibility issues with FFmpeg-generated DSS files in edge cases. Testing with your specific software version is recommended before relying on this workflow at scale.
The DSS format with the ADPCM IMA OKI codec does not expose adjustable bitrate or quality parameters through FFmpeg — the codec operates at a fixed encoding mode determined by the format specification. Unlike AAC or MP3, where you can specify -b:a to control bitrate, ADPCM IMA OKI has no such knob in FFmpeg's implementation. The output quality is therefore fixed and cannot be tuned from the command line.
On Linux or macOS, you can loop over all FLV files in a directory with: for f in *.flv; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.flv}.dss"; done. On Windows Command Prompt, use: for %f in (*.flv) do ffmpeg -i "%f" -c:a adpcm_ima_oki "%~nf.dss". Each file is processed sequentially, with the output DSS file taking the same base name as the input FLV.
Yes — if the FLV file contains no audio track, FFmpeg will produce an error or an empty DSS file because there is no audio stream to encode. FLV files that contain only video (or silent video) cannot be meaningfully converted to DSS. You can check whether your FLV has an audio stream by running ffmpeg -i input.flv in a terminal and looking for an audio stream in the output.
Technical Notes
The DSS format was developed jointly by Olympus, Philips, and Grundig as a proprietary standard for portable digital dictation recorders, and the ADPCM IMA OKI codec at its core is optimized for a narrow telephony-grade frequency range — typically up to around 3.4 kHz — which explains why DSS files sound acceptable for speech but poor for music or broadband audio. FLV source files commonly carry AAC audio at 44.1 kHz or 48 kHz; all of that high-frequency detail is lost during the transcode to ADPCM IMA OKI. FFmpeg does not preserve any metadata from the FLV container (such as title, author, or duration tags) in the DSS output, as the DSS format has a proprietary and largely undocumented header structure. There is no lossless path to DSS — all conversions are lossy. If your FLV file contains multiple audio tracks (rare, but possible), only the first audio stream will be encoded into the DSS output, as DSS does not support multiple audio tracks.