音频数字滤波器设计与实现论文.docVIP

音频数字滤波器的设计与实现 学院名称： 专 业： 班 级： 姓 名： 指导教师姓名： 指导教师职称： 2012年 6 月 音频数字滤波器的设计与实现 摘要：随着现代电子系统的发展，数字滤波器成为数字信号处理的重要组成部分。在家庭中和专业音频设备等一些领域，数字音频处理技术已经慢慢地取代模拟音频处理技术。音频处理的数字化，是利用数字信号处理算法对收集到的音频信号进行处理来实现的。 本文介绍了一个基于FPGA的音频数字滤波器的设计方案。该设计方案以最小均方（LMS）算法为核心，并分析推导了LMS算法公式。介绍了利用LMS算法的FIR自适应数字滤波器，并说明了该结构滤波器的特点。利用System Generator软件建立算法的FPGA硬件模型。利用前一时刻已获得的滤波器参数去自动的调节现时刻的滤波器参数，当输入信号的统计特性未知，或者输入信号的统计特性变化时，利用自适应滤波器可以实现最优滤波。仿真实验结果表明，该滤波器可实现对加噪后的音频信号自适应滤波。 关键词：数字滤波器；自适应滤波器；LMS；FIR；FPGA Audio Digital Filter Design And Lmplementation Abstract：With the development of modern electronic systems, the digital filter has become an important part of digital signal processing. In the family and the professional audio equipment and other fields, digital audio processing technology is slowly replacing the analog audio processing technology. The digital audio processing is the use of digital signal processing algorithm for the collected of audio signal processing to achieve. This paper introduces a FPGA based audio digital filter design scheme. The scheme takes the least mean square (LMS) algorithm as the core. LMS algorithm formula is derived in the paper. The LMS algorithm is used in FIR adaptive digital filter and the structure characteristics of the filter is introduced. We use System Generator software establish the FPGA hardware model of the filter. The previous filter’s parameters are used to adjusted present filter’s parameters automatically. When input signals’ statistical characteristics is unknown or time-varying, the adaptive filter can be used to realize optimal filtering. The simulation results show the filter can realize adaptive filtering of audio signal polluted by the noise. Keywords：digital filter；adaptive filter；LMS；FIR；FPGA 目录 前言