Convert MTS to DSS — Free Online Tool
Convert MTS camcorder footage (AVCHD with H.264 video and AC-3/AAC audio) to DSS format, extracting and re-encoding the audio track into the ADPCM IMA OKI codec used by Olympus and Philips digital dictation devices. This tool runs entirely in your browser — no upload required.
to
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
MTS files are AVCHD recordings from Sony or Panasonic camcorders, packaged in an MPEG-2 Transport Stream container with H.264 video and typically AC-3 or AAC audio. DSS (Digital Speech Standard) is a purely audio-only format — it has no video track and uses ADPCM IMA OKI, a low-bitrate codec optimized for voice intelligibility on digital dictation hardware. During this conversion, the H.264 video stream is completely discarded, and the audio is decoded from its original AC-3 or AAC encoding and then re-encoded into ADPCM IMA OKI at a fixed, low sample rate (8 kHz mono). This is a full transcode of the audio, not a stream copy, meaning some audio quality loss occurs. The resulting DSS file is a compact, speech-optimized audio file compatible with Olympus, Philips, and Grundig dictation devices and transcription software.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which is running here as FFmpeg.wasm compiled to WebAssembly — the same command works identically on your local desktop installation of FFmpeg. |
-i input.mts
|
Specifies your source MTS file as the input — an AVCHD MPEG-2 Transport Stream from a Sony or Panasonic camcorder, typically containing H.264 video and AC-3 or AAC audio. |
-c:a adpcm_ima_oki
|
Instructs FFmpeg to encode the audio track using the ADPCM IMA OKI codec, which is the specific low-bitrate speech codec required by the Digital Speech Standard (DSS) format used in Olympus and Philips dictation devices. The original AC-3 or AAC audio from the MTS file is fully decoded and re-encoded into this format. |
output.dss
|
Sets the output filename with the .dss extension, which tells FFmpeg to use the DSS muxer. The video stream from the MTS input is automatically dropped because the DSS container supports audio only. |
Common Use Cases
- Extracting spoken commentary or interview audio from a camcorder MTS recording to load into Olympus or Philips dictation transcription software for text transcription
- Archiving verbal field notes recorded on a Sony or Panasonic camcorder into the DSS format expected by a corporate dictation workflow or legal transcription system
- Stripping the audio from an AVCHD conference or lecture recording and converting it to DSS for playback on a standalone digital dictation device with no video capability
- Preparing speech recorded incidentally on a camcorder (e.g., witness statements, on-site voice memos) for import into dictation management platforms that only accept DSS files
- Reducing file size of a large MTS recording to a minimal DSS audio file when only the spoken words matter and video and high-fidelity audio are unnecessary
Frequently Asked Questions
DSS uses the ADPCM IMA OKI codec, which is specifically engineered for speech intelligibility at very low bitrates — not for music or high-fidelity audio. The codec operates at 8 kHz mono, which captures spoken voice clearly but will make music, ambient sound, or stereo content from your MTS recording sound noticeably degraded and flat. If your MTS file contains a clear voice recording, the DSS output will be quite usable for transcription purposes. If it contains music or rich environmental audio, the result will sound thin and lo-fi.
The H.264 video stream in your MTS file is completely dropped during conversion. DSS is an audio-only format with no support for video tracks, so FFmpeg discards the video entirely and only processes the audio. You will end up with a DSS file that contains only the audio from your camcorder recording. If you need to keep the video, you should use a different output format.
Yes, FFmpeg will decode the AC-3 audio from your MTS file and re-encode it into ADPCM IMA OKI for the DSS output. However, because DSS is mono at 8 kHz, any surround or stereo information in the AC-3 track will be down-mixed and heavily reduced in fidelity. The conversion will complete without errors, but the spatial audio characteristics of your original recording will be entirely lost.
No — DSS with ADPCM IMA OKI encoding has no configurable quality or bitrate parameters within FFmpeg. The codec operates at a fixed rate determined by the format specification, which is why there are no -b:a or -q:a flags in the conversion command. This is by design: DSS was built for standardized, low-bitrate dictation playback on dedicated hardware, not for flexible audio encoding.
You can batch process multiple MTS files using a shell loop. On Linux or macOS, run: for f in *.mts; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.mts}.dss"; done. On Windows Command Prompt, use: for %f in (*.mts) do ffmpeg -i "%f" -c:a adpcm_ima_oki "%~nf.dss". This runs the same conversion command on each file sequentially, producing one DSS file per MTS input. The browser-based tool processes one file at a time, so the command line is the best approach for bulk conversion.
No. MTS files embed metadata in the MPEG-2 Transport Stream structure — including recording timestamps and device information from Sony or Panasonic camcorders — but the DSS format has virtually no metadata support for custom fields. FFmpeg will not carry over MTS metadata tags into the DSS file. If preserving recording timestamps or source device information is important, you should document that data separately before converting.
Technical Notes
The MTS-to-DSS conversion is an extreme format reduction: you are going from a broadcast-grade AVCHD container carrying H.264 video at typical bitrates of 17–24 Mbps down to a purely mono speech codec running at roughly 8 kHz. The ADPCM IMA OKI codec used in DSS is a variant of Adaptive Differential Pulse-Code Modulation specifically tuned for the frequency range of human speech (approximately 300–3400 Hz), making it highly efficient for dictation but entirely unsuitable for wideband audio. One known limitation is that FFmpeg's DSS muxer has limited compatibility verification against all DSS-compliant playback devices; while the adpcm_ima_oki codec is correct for the format, some older or proprietary dictation hardware may have firmware-level restrictions on files not generated by their own software. The output file will have no video stream, no subtitle track support, and no chapter markers. Multiple audio tracks in the source MTS file are not preserved — only the default or first audio stream is processed. There are no quality-tuning options available for this output format, making the FFmpeg command fixed regardless of your input quality.