Earticle

초록

영어

Removal of noises from real-time speech signal is a typical problem. The signal interference initiated by background noise is a major problem in voice communication systems. Adaptive Filtering methods have emerged as an important technology for communication systems. This technique has been employed to improve the quality of the speech signal by cancelling the undesirable phenomenon such as acoustic noise. In this paper, for the removal of additional noise from speech signal an adaptive filter has been designed using LMS, NLMS, SLMS and VSS-LMS algorithms. This paper presents the instigation of Least Mean Square algorithm (LMS), Normalized Least Mean Square algorithm (NLMS), Sign Least mean square algorithm (SLMS) and Variable step size (VSS) algorithm on an infinite impulse response (IIR) filter using adaptive filter toolkit of LabVIEW software. User interface is designed using LabVIEW to obtain the learning curves for these adaptive algorithms. The final results show the comparison of the performance of the entire proposed algorithms with each other. The complete performance of the designed system in terms of stability and convergence rate has been observed.

목차

Abstract
 1. Introduction
 2. Background of Adaptive Filters
 3. Implementation of LMS Algorithm
 4. Implementation of NLMS Algorithm
 5. Implementation of Sign LMS algorithm
 6. Implementation of VSS LMS Algorithm
 7. Implementation of Learning Curves
 8. Simulation Results
 References

키워드

저자

• Divya Sharma [ School of Electronics and Electrical Engineering, Chitkara University, Punjab, India ]
• Rashpinder Kaur [ School of Electronics and Electrical Engineering, Chitkara University, Punjab, India ]
• Gurjinder Singh [ School of Electronics and Electrical Engineering, Chitkara University, Punjab, India ]

참고문헌

발행기관

간행물

표지보기 관심저널 등록

• 간행물명

  International Journal of Hybrid Information Technology

• 간기

  격월간

• pISSN

  1738-9968

• 수록기간

  2008~2016

• 십진분류

  KDC 505 DDC 605