Earticle

초록

영어

In this paper, model-based design is used to complete the design of an adaptive filter by Least Mean Square (LMS) algorithm, which implements the recovery process of audio signal disturbed by noise. We can quickly build a system simulation model by model-based design approach, and accomplish efficiently the system test, simulation and implementation. Theoretical analysis and experimental results show that the method of model-based design is not only valuable to the design and implementation of DSP system, but also can significantly improve the design efficiency of the DSP system.

목차

Abstract
 1. Introduction
 2. The Workflow of Model-Based Design
 3. The Introduction of LMS Algorithm
  3.1. The Principle of Adaptive Filter
  3.2. The LMS Algorithm
 4. The Design of Noise-Cancel of Audio Adaptive Filter
 5. Conclusion
 References

키워드

저자

• Hua Ge [ Department of Physics, Jining Normal University, Inner Mongolia, China ]
• Yang Nie [ Department of Physics, Jining Normal University, Inner Mongolia, China / Digital Engineering Center, Communication University of China, Beijing, China ]
• Lili Jing [ Department of Physics, Jining Normal University, Inner Mongolia, China ]
• Pengyu Zhao [ Department of Physics, Jining Normal University, Inner Mongolia, China ]

참고문헌

발행기관

간행물

표지보기 관심저널 등록

• 간행물명

  International Journal of Signal Processing, Image Processing and Pattern Recognition

• 간기

  격월간

• pISSN

  2005-4254

• 수록기간

  2008~2016

• 십진분류

  KDC 505 DDC 605