Using parallel signal processing in real-time audio matrix systems

  • Authors:
  • Jiri Schimmel

  • Affiliations:
  • Department of Telecommunications, FEEC, Brno University of Technology, Brno, Czech Republic

  • Venue:
  • WSEAS Transactions on Computers
  • Year:
  • 2010

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Abstract

The paper deals with design and performance analysis of algorithms that utilize parallel signal-processing methods and SIMD technology for multiply-and-add algorithm for digital audio signal processing. This algorithm is used for summing the gained input signals on output buses in applications for distributing, mixing, effect-processing, and switching multi-format digital audio signal in an audio signal network on desktop processors platforms. The subjective evaluation of latency caused by principle of the real-time digital audio processing is also studied in the paper Results of an analysis of speed-up and real-time performance of several summing algorithms are presented in the paper as well as subjective evaluation of the latency depending on the audio buffer size.