Digital Audio Codecs for Professional Recording and Broadcasting

SystemBase Digital Audio Codecs


The C400xr and C310xr digital audio codecs, have been designed and manufactured by Systembase to deliver unparalleled performance for professional, real-time audio applications. Both models incorporate the fast apt X100 Sub Band ADPCM compression system with a coding delay of only 3.8ms.

Additional design enhancements based upon the use of a 20 bit conversion process, have extended the audio bandwidth by almost 900Hz, resulting in a much richer natural sound.


Systembase codecs incorporate both an encoder and a decoder, to provide a real-time, full duplex audio circuit. The short coding delay of 3.8ms at 32KHz sampling rate, makes them suitable for bi-directional applications such as outside broadcasts (OB’s), studio to transmitter links (STL’s) and remote voice-overs. In addition, the integral ISDN inverse multiplexer, used to synchronise up to four ISDN B Channels, introduces a delay of just 0.5ms.


Systembase codecs allow cascade or tandem coding of audio with minimal degradation in quality. This feature is essential when contributing to Digital Audio Broadcast (DAB) radio or television stations using MPEG technology. In both cases, the use of psycho-acoustic compression, limits the number of times the audio signal can be encoded and decoded with a similar type of algorithm, before audible artefacts occur. This problem is overcome by Systembase codecs, which introduce a minimal I.7dBm of quantisation noise following each encode/decode cycle.


In the event of an ISDN call failing, the codecs will automatically drop down to provide the next best available service with the remaining connected calls. The user can configure the codecs to prioritise between bandwidth or stereo image. When operating in Mono, they have the capability of monoising the left and right input audio channels, and provide a pseudo stereo service to both the left and right outputs. In Dial-on-Audio, Trigger Dial or ISDN Backup modes, the codecs will automatically re-dial the ISDN lines if a call fails or disconnects.


Remote management facility is provided as standard, and enables full remote control and re-configuration via a computer. Multiple units can easily be linked together using a simple daisy-chain topology, and controlled centrally from a single computer. There are no limitations on the number of units that may be connected together. A codec unit at a remote or unattended site can readily be controlled by using standard analogue modems.


  • World-wide ISDN Operation (13 International ISDN Standards on-board)
  • 22.7KHz Stereo Over X.21 at 384Kbps
  • 15.6KHz Stereo Over ISDN 4 x 64Kbps
  • Real-time Applications – 3.8ms Coding Delay
  • 22.7KHz Mono Over ISDN 3 x 64Kbps
  • 2 x 3.9KHz Mono Over ISDN 64Kbps
  • 20 Bit A/D & D/A Conversion
  • AES/EBU Digital Interface with Sample Rate Conversion
  • 2.4 / 4.8 / 9.6 Kbps Auxiliary Data Port
  • 20 Bit VU Meters
  • Bi-directional Operation
  • Full Remote Management Facility
  • External Trigger Dial of ISDN (Contact Closure input)
  • Dial on Audio / Hang on Silence (Dial ISDN on Detection of Audio)
  • Studio to Transmitter Link with Fully Automatic Backup/Restore on ISDN
  • Selectable input monoising facility with left/right output duplication
  • 26 Configuration Memories / Stores
  • 1U Compact Rackmount Enclosure
  • Auto-ranging power supply
  • All Features Supplied as Standard

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