Convert ALAC to DSS — Free Online Tool
Convert ALAC audio files (.m4a) to DSS format using the ADPCM IMA OKI codec — the compressed speech encoding used in Olympus, Philips, and Grundig digital dictation devices. This tool runs entirely in your browser with no file uploads required.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
Drop your ALAC 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
ALAC (Apple Lossless Audio Codec) stores audio in an MPEG-4 container with mathematically perfect reconstruction of the original waveform — no audio data is discarded during compression. Converting to DSS is a lossy transcoding operation: FFmpeg decodes the ALAC stream to raw PCM audio, then re-encodes it using the ADPCM IMA OKI codec, which is a 4-bit adaptive delta pulse-code modulation scheme optimized for low-bitrate speech. DSS is a fixed, narrow format designed for dictation workflows — it operates at a low sample rate (typically 8000 Hz mono) and discards the full-spectrum fidelity of the ALAC source. The result is a compact file suitable for dictation playback devices, but the high-quality audio information present in the original ALAC recording cannot be recovered.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool, which handles the decoding of the ALAC-encoded MPEG-4 input and re-encoding into the DSS container with ADPCM IMA OKI audio. |
-i input.m4a
|
Specifies the input file — an MPEG-4 container (.m4a) holding a losslessly compressed ALAC audio stream, the native format used by Apple devices and iTunes for lossless audio storage. |
-c:a adpcm_ima_oki
|
Selects the ADPCM IMA OKI encoder for the audio stream, which is the codec required by the DSS format specification and used by Olympus, Philips, and Grundig digital dictation hardware and software. |
output.dss
|
Defines the output filename with the .dss extension, signaling FFmpeg to use the DSS muxer, which packages the ADPCM IMA OKI-encoded audio into the Digital Speech Standard container at 8000 Hz mono. |
Common Use Cases
- Transferring a voice memo or meeting recording made in Apple Voice Memos (saved as ALAC .m4a) into a Philips or Olympus digital dictation workflow that requires DSS input files
- Converting lossless audio archives of spoken interviews or oral history recordings into the DSS format for playback on legacy professional transcription foot-pedal systems
- Preparing Apple-ecosystem recordings for upload to dictation management software (such as Olympus Dictation Management System) that specifically ingests DSS files
- Reducing file storage footprint of large spoken-word ALAC recordings — such as lecture captures or legal dictation — where voice intelligibility is the only requirement
- Archiving transcribed dictation source files in DSS format to match the native output format of a digital recorder fleet, ensuring consistency across a document management system
Frequently Asked Questions
Yes, and the quality reduction is significant. ALAC is lossless and can faithfully reproduce full-fidelity stereo audio across the full audible frequency range. DSS using the ADPCM IMA OKI codec is aggressively optimized for low-bitrate speech intelligibility, operating at 8000 Hz mono — meaning stereo channels will be mixed down and all audio above roughly 4 kHz will be lost. For voice dictation and spoken word, the result is understandable but noticeably telephone-quality. Music or high-fidelity audio sources will sound degraded. This conversion is only appropriate when DSS device compatibility is the goal, not audio quality.
This scenario typically arises in professional environments where recordings are made on Apple devices or captured by high-quality recorders that export ALAC, but the downstream workflow — such as a medical, legal, or corporate transcription system — is built around Olympus or Philips DSS infrastructure. A doctor might record notes on an iPhone (ALAC .m4a output) and need to route those files through an existing dictation management platform that only accepts DSS. The ALAC source is incidental to the workflow requirement.
No. The DSS format does not support ID3-style metadata tags, chapter markers, or album artwork. Any metadata embedded in the ALAC .m4a container — including iTunes-compatible tags — will be stripped during the conversion. If preserving metadata matters, you should document it separately before converting, as there is no mechanism to store it within the DSS file structure.
The flag '-c:a adpcm_ima_oki' instructs FFmpeg to encode the audio stream using the ADPCM IMA OKI variant, which is the codec defined by the DSS format specification for digital dictation. DSS files are expected to contain this specific codec, so substituting a different audio codec (such as PCM or AAC) would produce a file that DSS-compatible devices and software cannot read. For valid DSS output, this codec selection is not optional — it is dictated by the format itself.
The single-file command shown converts one file at a time, but you can adapt it for batch processing in a shell script. On Linux or macOS, use: for f in *.m4a; do ffmpeg -i "$f" -c:a adpcm_ima_oki "${f%.m4a}.dss"; done. On Windows PowerShell: Get-ChildItem *.m4a | ForEach-Object { ffmpeg -i $_.FullName -c:a adpcm_ima_oki ($_.BaseName + '.dss') }. This is especially useful for converting an archive of Apple Voice Memos or iTunes-sourced recordings into a DSS dictation library.
ALAC is lossless compression that typically achieves only 40–60% size reduction from uncompressed PCM, and it preserves full stereo, high sample rate audio. DSS with ADPCM IMA OKI is designed for maximum compression of speech — encoding at 8000 Hz mono with 4-bit samples — resulting in bitrates around 12–16 kbps. By comparison, a typical ALAC file might encode at 700–1000 kbps for CD-quality stereo audio. The size difference is primarily due to the dramatic reduction in sample rate, channel count, and bit depth, not just codec efficiency.
Technical Notes
The DSS format was jointly developed by Olympus, Philips, and Grundig as a standardized container for professional digital dictation, and its audio encoding is tightly coupled to the ADPCM IMA OKI codec — a variant of IMA ADPCM adapted for the OKI semiconductor chipsets used in early digital recorders. FFmpeg's DSS muxer enforces an 8000 Hz mono constraint, so any ALAC source with a higher sample rate (44100 Hz, 48000 Hz, or 96000 Hz) or stereo/multichannel audio will be automatically downsampled and downmixed during encoding. This is expected behavior, not an error. One known limitation is that FFmpeg's DSS support is focused on decoding and basic encoding; some Olympus and Philips proprietary DSS variants (DSS Pro, DS2) may not be fully compatible with files produced by FFmpeg and could require dedicated Olympus or Philips transcription software for playback. If you encounter playback issues on physical dictation hardware, verify whether your device requires DSS or the newer DS2 format, as these are not interchangeable. There are no audio quality tuning parameters exposed for the ADPCM IMA OKI codec — bitrate and sample rate are fixed by the format specification.