基于MATLAB的FIR数字滤波器的设计

FIR数字滤波器是数字滤波器系统中常见的滤波器,本文介绍通过采用频率抽样设计法来设计FIR数字滤波器,利用MATLAB语言的强大功能和在数字滤波器设计上具有的独特优势完成设计中的计算与图形的绘制。对所设计的低通滤波器的仿真和频率特性分析表明,利用MATLAB来设计数字滤波器简单、方便。     

基于Matlab的IIR数字滤波器设计

介绍了基于Matlab的IIR数字滤波器设计方法。先确定数字滤波器的性能指标,再按照一定的映射规则（冲激响应不变法或双线性变换法）变换成模拟滤波器的性能指标,然后采用一定的逼近方法（巴特沃斯型或切比雪夫型）设计模拟滤波器,最后将模拟滤波器按照同样的映射规则转变成数字滤波器。同时介绍了设计IIR数字滤波器常用的Matlab函数。通过Matlab实验仿真,利用介绍的数字滤波器的设计方法,成功地设计出了满足预定指标的各型IIR数字滤波器。     

基于TMS320F2812的FIR数字滤波器设计

数字滤波器是现代数字信号处理的重要基础,文章阐述了数字滤波器的基本原理和利用TI公司提供的CCS软件(Code Composer Studio 2.21)进行数字滤波器设计的方法,并给出在CCS软件中实现的滤波器设计效果波形和实验波形。     

基于MATLAB的FIR数字滤波器设计实现

数字滤波器是数字信号处理中的重要组成部分。本文提出一种数字滤波器的设计方法——基于MATLAB的数字滤波器的原理与实现。文章通过一FIR滤波器设计实例说明如何利用MATLAB来完成数字滤波器的设计。     

基于TMS320F2812的FIR数字滤波器设计

数字滤波器是现代数字信号处理的重要基础,文章阐述了数字滤波器的基本原理和利用TI公司提供的CCS软件(Code Composer Studio 2.21)进行数字滤波器设计的方法,并给出在CCS软件中实现的滤波器设计效果波形和实验波形.     

分布式应变采集系统数据异常分析研究

正本文针对某型飞机的应变数据出现的"台阶"现象,基于对分布式应变采集系统结构、原理及滤波器逻辑设计的详细分析,得出系统的数字滤波器选型差异导致故障现状的出现,通过将"简单数据抽取+FIR(有限长单位冲击响应)数字滤波器"的数字滤波模式更改为"CIC数字滤波器(级联积分梳妆数字滤波器)+IIR数字滤波器(无限冲击响应数字滤波器)补偿,并修改某采集计数器跳转条件后,"台阶"故障现象消失,同时对修改后的数字滤波器做了详细的动态特性验证测试,测试     

FIR数字滤波器的仿真与实现

数字滤波器是数字信号处理中最重要的组成部分之一,在许多科学技术领域中,广泛使用滤波器对信号进行处理。本文介绍了FIR数字滤波器的设计方法,并利用Matlab实现了对FIR数字滤波器的仿真。仿真结果验证了该方法的正确性和可行性,达到了预期的效果。     

基于MATLAB的高通滤波器设计

为了进一步优化滤波器设计,便于及时调整滤波器的参数,提出了基于MATLAB平台利用滤波器设计与分析工具箱设计数字滤波器。阐述了数字滤波器的基本原理和滤波器设计的操作步骤,并进行仿真模拟。结果表明:所设计的滤波器可以有效抑制干扰信号,提高信噪比;设计过程快速简单,方便根据实际需求调整滤波器的参数,适用于工程实践中。     

基于MATLAB的FIR数字滤波器的设计

为了简化滤波器设计的过程,方便调整滤波器的参数,提出了基于MATLAB平台中滤波器设计与分析工具箱设计FIR数字滤波器。阐述了数字滤波器的基本原理和滤波器设计的详细步骤,并进行仿真。结果表明:所设计的滤波器可以有效抑制干扰信号,提高信噪比;整个设计过程简单,易于根据需求调整滤波器的参数,适用于工程实践。     

浅谈数字滤波器

本文通过基本数字运算,直观的阐述了数字滤波器的形成过程,并对数字滤波器的使用器件做了简要介绍。     

用滤色片来改善显色性

阐述如何用滤色片来改善显色性,基于相关理论,然后付诸实践,以一个白光LED为例,寻找与其相配的滤色片,以获得优良的显色性和较少的光损失。     

A continuous finite-time convergence fixed-lag FIR smoother using multiple IIR filters

In this paper, we propose a continuous finite-time convergence finite impulse response (FIR) fixed-lag smoother using multiple, or more than two, computationally efficient IIR filters. We describe the optimal design to improve and further optimize an existing scheme based on two IIR filters. Multiple IIR filters are utilized to minimize the estimation error variance of the proposed smoother under the condition that its estimate converges to a real state in a finite time. As the number of adopted IIR filters increases, the proposed smoother improves and its performance approaches that of the heavy computational fixed-lag minimum variance FIR smoother. By choosing the appropriate number of IIR filters, we can balance the trade-off between improved accuracy and increased implementation costs. To realize the optimal design of IIR filters with the limited number of IIR filters, their gains are determined using a particle swarm optimization scheme. Numerical examples are used to show that with an increasing number of IIR filters, the estimation error variance decreases monotonically while guaranteeing finite-time convergence.     

Approximation based on orthogonal and almost orthogonal functions

In this paper, we define a class of almost orthogonal rational functions of Legendre type in a new manner. Relations of these functions with classical exponentional functions orthogonal over interval (0, ∞), as well as classical polynomials orthogonal over (0, 1) are explained. Defining relations of these functions can be used for designing almost orthogonal filters. These filters are generators of orthogonal signals and can be successfully applied in finding the best signal approximation in the sense of the mean square error. The filters orthogonal property enables building of physical (in this case electrical) models of dynamical systems (the sources of signals to be approximated) either with less components for the same model accuracy or higher accuracy for the same number of components than the other known models. New filters represent further improvement of previously designed filters, by the same authors, in the sense of simplicity, higher accuracy, lesser approximation time and even a possibility to approximate signals generated by systems with built-in imperfections. Series of experiments were performed to analyze the dependence of approximation accuracy and the number of filters sections.     

A special class of multiplierless digital filters

This paper deals with the design of discrete low-pass filters, using a special class of polynomials of the second and third order, with coefficients which are powers of two. To take advantage of the characteristics of these polynomials, the discrete integrator is used as a basic component. The filters so derived are multiplierless resulting in hardware economy. Simulation indicates that the filters obtained are equally good or better than those obtained by the application of the impulse invariance method.     

IIR数字陷波器的设计及FPGA实现

IIR（无限冲激响应）数字滤波器在许多领域得到广泛应用.这里介绍了一种利用Matlab设计一个IIR数字陷波器的方法,接着在硬件实现时将其转化为二阶级联形式,以VerilogHDL语言书写模块,最后利用Altera公司的Quartus Ⅱ软件进行FPGA设计及仿真.     

适用于海事卫星终端的体声波滤波器设计

近年来,国际海事卫星应用范围的扩展对终端设备的微型化与抗噪性提出了更高的要求,薄膜体声波滤波器成为人们关注的焦点。相对于传统滤波器,体声波滤波器具有更高的谐振频率和温度稳定性,更低的损耗和成本,同时可以实现芯片集成。本文介绍了FBAR体声波滤波器的工作原理与设计模型,在此基础上设计出了适用于BGAN下行频段的窄带滤波器。为实现材料参数的实时调整,本文采用Mason模型,通过改变串并联单元级数与面积比,最终设计出滤波器频带为1525~1559Mhz,带外抑制60dB,插损4.3dB。     

Selectivity nomographs for classical filters

Nomographs for determining the filter order of classical filters based on selectivity requirements are presented. The selectivities for a variety of standard classical filters are summarized in equation form and the general selectivity nomograph is constructed. The selectivity equations are then converted into nomograph form by applying the relationship between the transfer function and the response slope. Design examples are presented to demonstrate the usefulness of the selectivity nomographs. These nomographs can be used to gauge filter performance and combined with optimization techniques can yield superior classical filter designs.     

A class of fast and accurate deterministic trend decomposition in the spectral domain using simple and sharp diffusive filters

This study proposes a spectral domain algorithm to remove the deterministic non-periodic trend from a time series using a class of fast, sharp and diffusive filters. These filters are principally the iterative moving least squares methods weighted using Gaussian windows. The responses of the filters expressed in analytic forms are proven to be diffusive. If it is a polynomial of finite degree, the embedded trend can be decoupled by the filters with specific order and iteration steps. The filters’ order, transition zone, error tolerance, iteration number and smoothing factor are subject to two algebraic equations to form a specific class. The operation counts of all filters are slightly larger than twice that applying a Fast Fourier Transform (FFT). It is numerically shown for a given transition zone and tolerance, there is a filter generating the shortest error penetration distance among all the filters. If either the trend or a spectral band is the main concern, there is an optimal strategy to shrink the error penetration distance. The numerical results show the filter has better performance than several existing methods. In addition, four examples successfully show direct applications of the filter’s response.     

Pulse nonstationary processes generated by dynamic systems with random structure

Dynamic system with a random structure described by a set of the first-order stochastic differential equations (SDE) is used as a generating model of nonstationary pulse stochastic processes. Physically the system presents the combination of the so-called partial filters related to the isolated states of the considered process, switched by a Poissonian point process and excited by a vector delta-correlated stream of impulses with the randomly distributed energy. The filters’ outputs are components of the vector Markov continuous-jump process with statistics depending on the partial SDEs operators, intensity of switching process and distributions of the exciting impulses’ energies. The approach proposed was used as a simulation model of the Middleton “Class-A “generally non-Gaussian noise. The results demonstrate that the main features of statistical characteristics of the noise envelope are reproduced rather well with the help of a bistate system with random structure.     

On the construction of transitional filter nomographs

The construction of nomographs for transitional classical filters is described. Gain functions of classical filters are related to filter requirements resulting in a formulation for the general gain nomograph. The transitional filters that are products of approximating polynomials are incorporated into the general gain nomograph resulting in transitional filter nomographs that are sums of the individual nomographs. Nomographs for transitional filters using alternative forms where poles are interpolated are also considered. The resulting nomographs allow for quick optimization of transitional filter frequency response in many cases. Design examples are submitted and discussed. The proposed transitional filter nomographs provide the engineer with increased insight into the selection of classical transitional filters with optimum frequency response.