基于Matlab的IIR数字滤波器设计

数字滤波是数字信号处理的重要内容,可分为FIR和IIR两大类。文章介绍了基于MATLAB的IIR数字滤波器设计方法。先确定性能参数,再按照映射规则(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参数,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映射规则转变成数字滤波器。通过Matlab实验仿真,成功地设计出了满足预定指标的IIR数字滤波器。     

基于FPGA的IIR数字滤波器的实现

数字信号处理在科学和工程技术许多领域中得到广泛的应用,与FIR数字滤波器相比,IIR数字滤波器可以用较低的阶数获得较高的选择性,本文采用一种基于FPGA的IIR数字滤波器的设计方案,首先分析了IIR数字滤波器的原理及设计方法,然后通过MAX＋PLUSⅡ的设计平台,采用自顶向下的模块化设计思想将整个IIR数字滤波器分为：时序控制、延时、补码乘加和累加4个功能模块。分别对各模块采用VHDL进行描述后,进行了仿真和综合。仿真结果表明,本课题所设计的IIR数字滤波器运算速度较快,系数改变灵活,有较好的参考价值。     

基于FPGA的罗兰C前端IIR数字带通滤波器设计

针对罗兰C前端带通滤波的需求,提出了采用级联形式在FPGA上实现罗兰C数字带通滤波器的方法。首先利用Matlab设计出满足要求的滤波器,考虑硬件设计要求将参数进行取整,并对取整前后的滤波效果进行了比对分析。在硬件程序设计前在Matlab下用级联式差分方程模拟硬件滤波算法,以提高设计的成功率。最后在FPGA下用Veril...     

基于单自由度振动模型的IIR数字滤波器设计

无限冲击响应(IIR)数字滤波器设计通常需要先设计S域滤波器,再变换到Z域。为简化IIR数字滤波器的设计过程,本文提出利用单自由度系统(SDOF)模型直接设计IIR数字滤波器的方法,包括低通、带通、高通、带阻等。仿真结果表明设计过程直接、简单,滤波器参数不需要随采样频率而变,具有良好特性,是一种设计IIR数字滤波器的简便易用的新方法。     

基于Matlab的IIR数字滤波器设计及DSP实现

IIR滤波器是一种被广泛应用的基本的数字信号处理部件.基于DSP信号处理的优越性,将Matlab与DSP相结合应用于IIR滤波器的设计.介绍了IIR数字滤波器的理论及其Matlab常用设计函数.并针对TI公司的TMS320VC5416 DSP,结合某高通滤波器的设计,给出了其Matlab仿真设计及在DSP上的实现过程及结果.该方法具有较强的实用性,对其它数字滤波器设计及DSP实现提供了参考价值.     

基于人工鱼群算法的IIR数字滤波器设计

将人工鱼群算法（AFSA）用于IIR数字滤波器设计，建立了相应的优化模型，给出了简化的人工鱼群算法及其实现步骤。最后，将该算法用于低通、带通IIR数字滤波器的设计，并与粒子群算法进行了比较。仿真结果证明了AFSA的有效性，并且具有算法灵活、简单，全局收敛性好。收敛速度快的优点。     

基于最小二乘滤波器的空域视频掩盖算法

针对无线信道传输过程中可能出现的单个或连续宏块丢失现象,提出了一种有效的空-域差错掩盖算法.算法根据丢失像素点中是否含有边界,把丢失的像素点分为平稳点和非平稳点.对于不含有边缘的平稳点,采用线性插值的算法;对于含有边界的非平稳点,建立最小二乘滤波器(LS)模型,通过多方向的扫描方式得出预测值.实验结果表明,算法能够获得优秀的图像质量,在降低了复杂度的同时,适用于视频图像的实时传输.     

基于免疫算法的IIR数字滤波器优化与实现

由于IIR数字滤波器设计实质上是一个非线性高维复杂函数优化问题,文中提出基于具有全局搜索能力强,收敛速度快特点的免疫算法实现IIR数字滤波器优化设计的新方法,给出了IIR滤波器优化设计的数学模型,描述了应用免疫算法优化设计IIR数字滤波器的具体实现步骤。通过低通和高通IIR数字滤波器设计的仿真结果表明方法的有效性和高效性。     

高阶IIR滤波器的FPGA实现

结合IIR数字滤波器的基本结构,针对分布式算法中查找表规模过大的缺点,采用级联或并联结构,利用多块查找表使得硬件规模极大地减小,提出了并行和串行相结合的设计方案,并且实现了级联方式的10阶IIR低通滤波器。通过试验验证了该方法的有效性和实时性。     

Design of IIR digital filters based on eigenvalue problem

This paper presents a new method for designing IIR digital filters with optimum magnitude response in the Chebyshev sense and different order numerator and denominator. The proposed procedure is based on the formulation of a generalized eigenvalue problem by using Remez exchange algorithm. Since there exist more than one eigenvalue in the general eigenvalue problem, we introduce a very simple selection rule for the eigenvalue to be sought for where the rational interpolation is performed if and only if the positive minimum eigenvalue is chosen. Therefore, the solution of the rational interpolation problem can be obtained by computing only one eigenvector corresponding to the positive minimum eigenvalue, and the optimal filter coefficients are easily obtained through a few iterations. The design algorithm proposed in this paper not only retains the speed inherent in the Remez exchange algorithm but also simplifies the interpolation step because it has been reduced to the computation of the positive minimum eigenvalue. Some properties of the filters such as lowpass filters, bandpass filters, and so on are discussed, and several design examples are presented to demonstrate the effectiveness of this method     

IIR数字滤波器的研究与设计

文中探讨了利用模拟滤波器设计IIR数字滤波器过程中的转换方法,脉冲响应不变法和双线性变换法,在MATLAB中以两种方法设计了数字巴特沃思低通滤波器。探讨了MATLAB中非低通数字滤波器的完全工具函数设计法和分步函数设计法,以上述方法分别设计了切比雪夫I型数字低通、高通、带通、带阻滤波器。     

Design of causal stable IIR perfect reconstruction filter banksusing transformation of variables

A generalisation to the design technique of Tay and Kingsbury (see IEEE Trans. Circuits Syst. II: Analog Digit. Signal Process., vol.43, no.2, p.274-79, 1996) for two-channel, causal stable IIR perfect reconstruction filter banks is presented based on transformation of variables. Previously the transformation functions used were allpass, but this yielded subband filters with a fairly large overshoot in their frequency design responses. By relaxing the requirement of using allpass transformation functions, filters with improved response (lower and no overshoot) are achievable. Several design examples are presented to show the flexibility of the design technique     

Approximate realization of digital fractional forward operator using digital IIR filter

In this paper, a simple and an efficient approach for approximating the digital fractional forward operator z ^m (0?<?m?<?1) using digital infinite impulse response (IIR) filter is proposed. In this method, the coefficients of the closed form digital IIR filter derived for the approximation of the fractional forward operator, in a given frequency band, are based on approximation of fractional order systems. First, analog rational function approximation, in a given frequency band, of the fractional power zero (FPZ) is given. Then, the forward difference generating function is used to obtain a closed form IIR digital filter equivalent of the continuous FPZ, which approximates the digital fractional forward operator. Finally, illustrative examples have been presented to illustrate the effectiveness of the proposed design technique of the fractional forward operator z ^m approximation and its use in performing a fractional m-step prediction.     

基于MATLAB的FIR滤波器的设计

基于MATLAB信号处理工具箱,设计一种滤波器,针对信号的分离和滤除,要求有良好的幅频特性和线性相位特性,且可通过参数来改变其主瓣宽度和旁瓣衰减。提出基于窗函数法设计滤波器,加入信号源,利用FDATool设计和分析模块电路,经过Simulink仿真,对其进行时域、频域分析和滤波。实验结果表明,本设计方法能够有效的滤除无...     

Nonlinear adaptive IIR digital filter for linear system modeling

The structure and adaptive algorithm of a nonlinear adaptive IIR digital filter is presented. It does not need stability monitoring in the adaptive process, which has always been a computational burden and disturbs the adaptive process in linear adaptive IIR filtering. The individual parameter adaptation scheme is incorporated into the adaptive algorithm to optimally adjust each parameter at every iteration to improve convergence speed. Simulation results are conducted for linear IIR system modeling     

Mapping 3-D IIR digital filter onto systolic arrays

We present here an efficient systolic implementation for 3-D IIR digital filters. The systolic implementation is obtained by using an algebraic mapping technique. This new mapping technique gives us the choice to mix pipelined variables and broadcast variables. We also determine, through the mapping method, the buffer sizes, the direction of variables propagations and the data feeding and extracting points. The resultant systolic array implementation is a modular structure composed of 2-D filter modules connected by simple buffers. This new systolic implementation is regular, modular and amenable to VLSI Implementation.     

Design of IIR digital notch filters

A method is presented for the design of notch filters with specified notch frequency ₀ and 3-dB rejection bandwidthB _t, using a first-order real all-pass filter, wherein the only coefficient is used to control the notch frequency. To control the bandwidth, use is made of a new amplitude change function (ACF), and it is shown that given notch filter specifications can be exactly met thereby. Also, using the ACF, it is shown that stability of the second-order notch filter designs can be improved along with the noise gain.     

An adaptive IIR filter algorithm based on observers

The output error approach to adaptive IIR filtering is considered from a state observation perspective, and a new algorithm, termed the observer-based regressor filtering (OBRF) algorithm, is developed. The convergence requirements of the OBRF are established as a persistent excitation condition on the regressor and a strict positive reality (SPR) condition on an operator arising in the algorithm. Speed of convergence experiments show that the OBRF algorithm converges more quickly than the related output error algorithm for the hyperstable adaptive recursive filter (HARF), although the OBRF algorithm converges as quickly as typical equation error schemes. The OBRF is shown to compare favorably with equation error with respect to parameter bias in the presence of output measurement noise. Thus, OBRF is a compromise between the equation error and output error approaches. In addition, algorithm parameter selection to satisfy the SPR condition for OBRF is explored and compared with the related conditions for HARF     

An 85-MHz fourth-order programmable IIR digital filter chip

The authors describe the design and VLSI implementation of a single-chip 85-MHz fourth-order infinite impulse response (IIR) digital filter chip fabricated in 0.9-μm CMOS technology. The coefficient and input data word lengths of the filter are 10 b each, and the output data word length is 15 b. The coefficients are fully programmable. The chip can be programmed to implement any IIR filter from first to fourth order or an FIR filter up to 16th order at sample rates up to 85 MHz. A total of seventeen 10×10 multiply-add modules are used in this chip. The chip contains 80000 devices in an active area of 14 mm². It dissipates 2.2 W at 85-MHz clock rate and performs over 1.5×10⁹ multiply-add operations per second. The underlying filtering algorithm, chip architecture, circuit and layout design, speed issues, and test results are described. The results of an E-beam probing experiment on packaged chips at 100-MHz clock rates are also presented and discussed