Convert FLAC to DSS — Free Online Tool
Convert FLAC lossless audio files to DSS (Digital Speech Standard) format using the ADPCM IMA OKI codec — a highly compressed, speech-optimized format used by Olympus, Philips, and Grundig digital dictation devices. This tool runs entirely in your browser with no file uploads required.
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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
FLAC stores audio as losslessly compressed PCM data, preserving every sample of the original recording. Converting to DSS transcodes the audio using the ADPCM IMA OKI codec, a form of Adaptive Differential Pulse-Code Modulation optimized for low-bitrate speech encoding. This is a significant lossy transformation: the full-fidelity wideband audio in your FLAC file is downsampled to a narrow frequency range suited for voice, dramatically reducing file size. The ADPCM IMA OKI codec encodes audio differences between samples rather than absolute values, which is highly efficient for the relatively predictable waveforms of human speech. Music or non-speech audio will sound noticeably degraded in the output DSS file.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles the decoding of the input FLAC file and re-encoding into the DSS output format. |
-i input.flac
|
Specifies the input file — a FLAC audio file encoded with the lossless FLAC codec. FFmpeg will fully decode this to uncompressed PCM internally before re-encoding. |
-c:a adpcm_ima_oki
|
Sets the audio codec to ADPCM IMA OKI, the proprietary codec used in DSS files. This lossy codec applies adaptive differential pulse-code modulation optimized for low-bitrate speech audio, which is the required codec for the DSS container. |
output.dss
|
Specifies the output filename with the .dss extension, which tells FFmpeg to use the DSS container format. The combination of this container and the adpcm_ima_oki codec produces a file intended for digital dictation devices and transcription software. |
Common Use Cases
- Transferring recorded interviews or voice memos stored as FLAC into a format compatible with Olympus or Philips digital dictation hardware and transcription software
- Archiving high-quality FLAC recordings of meetings or lectures in DSS format for long-term storage at minimal file sizes when audio fidelity is not critical
- Preparing FLAC voice recordings for import into dictation and transcription workflows that require DSS input, such as legal or medical transcription pipelines
- Converting FLAC captures from a high-quality microphone into a DSS file for playback on a dedicated portable dictation device that does not support FLAC
- Reducing storage footprint of a large collection of voice-only FLAC recordings by re-encoding them to the highly compact DSS format for archival or sharing with transcriptionists
Frequently Asked Questions
Yes, and significantly so. FLAC is lossless, meaning it perfectly preserves the original audio at full bit depth and sample rate. DSS uses the ADPCM IMA OKI codec, which is a lossy, low-bitrate codec designed specifically for narrow-band speech. Music, environmental sounds, or high-frequency content in your FLAC file will be heavily degraded or lost entirely. DSS is only suitable for voice recordings where intelligibility matters more than fidelity.
DSS files produced using FFmpeg with the adpcm_ima_oki codec may not be fully compatible with all proprietary dictation hardware or vendor-specific DSS software, as Olympus, Philips, and Grundig have each implemented slight variations of the DSS standard. For guaranteed device compatibility, it is best to test the output file on your specific device or use the manufacturer's own software for final encoding.
No. The DSS container format does not support the kind of rich metadata tagging that FLAC does via Vorbis comments. Information such as artist, album, title, and track number stored in your FLAC file will not be carried over to the DSS output. If metadata preservation matters, keep the original FLAC file as your archive copy.
DSS files are dramatically smaller than FLAC files. FLAC typically stores audio at 16-bit or 24-bit depth with sample rates of 44.1 kHz or higher, while the ADPCM IMA OKI codec in DSS targets very low bitrates optimized for speech, often below 16 kbps. For a voice recording, you can expect file size reductions of 90% or more compared to the source FLAC file.
Copy the FFmpeg command displayed on this page — 'ffmpeg -i input.flac -c:a adpcm_ima_oki output.dss' — and run it in a terminal after installing FFmpeg on your system. Replace 'input.flac' with the actual path to your file and 'output.dss' with your desired output filename. This command works identically on Windows, macOS, and Linux.
FFmpeg does not natively support glob-based batch conversion in a single command, but you can use a shell loop to process multiple files. On Linux or macOS, run: 'for f in *.flac; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.flac}.dss"; done'. On Windows PowerShell, use: 'Get-ChildItem *.flac | ForEach-Object { ffmpeg -i $_.FullName -c:a adpcm_ima_oki ($_.BaseName + ".dss") }'. This applies the same adpcm_ima_oki encoding to each file in sequence.
Technical Notes
The DSS format was developed jointly by Olympus, Philips, and Grundig as a proprietary container for digital dictation, and its primary codec — ADPCM IMA OKI — is an adaptive differential PCM variant tuned for speech-band audio, typically at 8 kHz sample rate. When FFmpeg encodes to DSS using adpcm_ima_oki, the input FLAC audio (which may be 44.1 kHz stereo or higher) will be downmixed and resampled to match the codec's narrow-band requirements, resulting in audio that covers only the frequency range needed for voice intelligibility (roughly 300 Hz to 3.4 kHz). There are no adjustable quality parameters for the adpcm_ima_oki codec — the bitrate and sample rate are fixed by the codec specification. FLAC features such as cue sheets, ReplayGain tags, and chapter markers have no equivalent in DSS and will be silently discarded. Because DSS is a proprietary format with multiple vendor-specific revisions (DSS and DSS Pro), FFmpeg's output may not be recognized by all dictation software or devices without additional compatibility testing.