Adaptive FIR Filter to Compensate for Speaker Non-linearity

  • Varsha Varadarajan
  • Kinnera Pallavi
  • Gautam Balgovind
  • J. Selvakumar
Conference paper
Part of the Smart Innovation, Systems and Technologies book series (SIST, volume 32)

Abstract

In this paper we have implemented an adaptive filter to compensate for the non-linearity in a speaker. An attempt has been made to minimize the Mean Square Error (MSE) and convergence time using the LMS adaptive algorithm. Two adaptations of the LMS have been considered, the general adaptive LMS algorithm and the Leaky LMS algorithm. The Leaky LMS adaptation is observed to be more efficient with almost a 40 % decrease in convergence time. The filter coefficients for the above objective function are obtained using MATLAB. The target processor for implementing the two algorithms is Tensilica/Xtensa SDK toolkit using ‘C’ language which enables the codes to be directly dumped on to hardware.

Keywords

Adaptive echo cancellation LMS adaptive algorithm Leaky LMS algorithm Variable leaky LMS Convergence rate 

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Copyright information

© Springer India 2015

Authors and Affiliations

  • Varsha Varadarajan
    • 1
  • Kinnera Pallavi
    • 1
  • Gautam Balgovind
    • 1
  • J. Selvakumar
    • 1
  1. 1.Department of ECESRM UniversityKattankulathurIndia

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