基于自适应滤波对音频信号的处理详解

毕业设计论文

题目基于自适应滤波对音频信号的处理

学生

完成日期2012年5月

基于自适应滤波对音频信号的处理

摘要

自适应滤波器是统计信号处理的一个重要组成部分。在信号的传播过程中，通常会受到噪声的干扰，这时候自适应滤波器就体现出了它的重要作用。传统的滤波器只能过

滤一种或几种干扰噪声，然而随着信号传输环境的不断变化，传统的滤波器已经不能适应，这就需要一种根据环境的改变，通过自适应算法来改变滤波器的参数和结构，来达到更好滤波效果的滤波器。自适应滤波器是利用前一时刻获得的滤波参数，自动地调节、更新现时刻的滤波参数，以适应信号和噪声未知的统计特性，从而实现最优滤波。

本文从自适应滤波器研究的意义入手，介绍了自适应滤波器的基本理论思想，具体阐述了自适应滤波器的基本原理、算法及设计方法。自适应滤波器的算法是整个系统的核心。因LMS算法具有低计算复杂度、在平稳环境中的收敛性好、其均值无偏地收敛到维纳解和利用有限精度实现算法时的稳定性等特性，使LMS算法成为自适应算法中应用最广泛的算法，所以最终采用基于LMS算法设计自适应滤波器。对读取一段wav格式的音频文件采用 MATLAB 进行仿真，通过实验结果来体现该滤波器可以根据信号随时修改滤波参数，达到动态跟踪的效果，使滤波信号更接近于原始信号。

关键词：自适应滤波；LMS算法； Matlab

ABSTRACT

The adaptive filter is an important part of the digital signal processing. in the spread of the signal process, usually subject to noise interference, and this was reflected when the adaptive filter out of its important role. The traditional filters can only filtration one or more of the noise interference, however, with the constant change of the signal transmission environment, the traditional filters can not adapt tochanges in the environment which requires a through adaptive algorithm tochange the filter parameters and structure of the device to reach the better the filtering effect of the filter. The adaptive filter is to use a time before get the filter parameters have been automatically adjust and update the current moment of filter parameters, to adapt to the signal and noise statistical properties of the unknown in order to achieve optimum filter.

This paper,from the adaptive filter the significance of research and introduced its elementary theory, algorithm and design method. The core of the whole system is the auto-adapted filter's algorithm. For LMS algorithm has low computational complexity, in the environment of steady convergence, the mean unbiased to converge to a wiener solution and the use of the limited precision of the stability of the algorithm and other characteristics .

LMS algorithm as adaptive algorithm in the application of the most a wide range of algorithms.，So Finally the design of adaptive filters based on LMS algorithm.To read a wav format audio files using MATLAB simulation and experimental results to reflectthe filter can be modified at any time according to the signal of the filter parameters to achieve the effect of dynamic tracking, so that the filtered signalcloser to the original signal.

Keywords: adaptive filter ; LMS algorithm ; Matlab

目录

第一章绪论 (6)

1.1 引言 (6)

1.2 自适应滤波理论的发展 (6)

1.3 自适应滤波器的基本概念 (7)

1.3.1 自适应滤波器特征及特点 (7)

1.4 自适应滤波器原理 (8)

1.5 课题研究意义和目的 (9)

第二章自适应LMS算法的研究 (11)

2.1自适应LMS算法的发展 (11)

2.1.1 LMS算法的历史 (11)

2.1.2 LMS算法的现状 (11)

2.1.3 LMS算法的发展前景 (11)

2.2 自适应LMS算法的原理 (12)

2.3 自适应LMS算法降噪的原理 (14)

第三章 WAV文件格式 (15)

3.1 WAV文件格式介绍 (15)

3.2 PCM数据的存放方式 (16)

3.3 WAV文件格式 (17)

第四章仿真实验 (19)

4.1 MATLAB语言介绍 (19)

4.1.1 MATLAB的特点 (19)

4.1.2 MATLAB的优势 (19)

4.2 MATLAB仿真 (20)

4.2.1 MATLAB程序仿真 (20)

4.2.2 仿真结果 (22)