共查询到19条相似文献,搜索用时 156 毫秒
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数字滤波器是一种用来过滤时间离散信号的数字系统,通过对抽样数据进行数学处理来达到频域滤波的目的。根据其单位冲激响应函数的时域特性可分为两类:无限冲激响应(IIR)滤波器和有限冲激响应(FIR)滤波器。与IIR滤波器相比,FIR的实现是非递归的,它总是稳定的,更重要的是,FIR滤波器在满足幅频响应要求的同时,可以获得严格的线性相位特性。因此,它在高保真的信号处理,[第一段] 相似文献
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数字滤波器在数字信号处理中具有举足轻重的作用。在分析IIR数字滤波器和FIR数字滤波器设计原理的基础上,在设计过程中使用MATLAB提供的GUI工具,实现了方便用户使用的数字滤波器交互界面开发,进行应用演示,最后对IIR数字滤波器和FIR数字滤波器进行了对比分析。 相似文献
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在有限冲激响应(Finite Impulse Response,FIR)滤波器设计中,如果系统只要求通带或某个频域区间具有线性相位而其他频域区间相位非线性,则系数对称的FIR滤波器设计方法不再适用。为此,提出了一种基于二阶锥规划(Second-Order Cone Programming,SOCP)的通带线性相位FIR滤波器设计方法。该方法使用二阶锥规划实现滤波器设计,其中优化目标为通带最小群延迟,约束条件为全频域振幅误差。实验结果显示,所提方法设计的FIR滤波器有着很好的幅频特性和通带线性相位,通带群延迟误差很小。该方法实现简单,计算复杂度低,可以广泛应用于数字信号处理领域。 相似文献
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基于Matlab的语音信号数字滤波 总被引:1,自引:0,他引:1
在Matlab软件平台上,对录制的语音信号采样,分析时域波形和频谱图。根据给定的指标,分别用窗函数法设计FIR数字滤波器和用双线性变化法设计IIR数字滤波器,并对语音信号进行滤波,去除噪声。通过分析滤波后信号的频谱图,简单而有效地阐述了两种数字滤波器在信号处理中的优势。 相似文献
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FIR滤波器具有绝对稳定性和线性相位的优势,然而当对滤波器的频域性能要求较高时,FIR滤波器通常需要很高的阶数,这使得FIR滤波器硬件执行的复杂度很高。为降低FIR滤波器的硬件执行复杂度,诸多研究者进行了探索。文章对低复杂度FIR滤波器设计方法进行研究,着重介绍比较典型的频率响应罩设计方法、外插脉冲响应设计方法和基于压缩感知的设计方法。 相似文献
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This paper first presents the fundamental principles of the microwave photonic filters.As an example to explain how to implement a microwave photonic filter, a specific finite impulse response (FIR) filter is illustrated.Next, the Q value of the microwave photonic filters is analyzed theoretically, and methods around how to gain high Q value are discussed.Then,divided into FIR filter, first-order infinite impulse response (IIR) filter, and multi-order IIR filter, several novel microwave photonic filters with high Q value are listed and compared.The technical difficulties to get high Q value in first-order IIR filter and multi-order IIR filter are analyzed concretely.Finally, in order to gain higher Q value, a multi-order IIR microwave photonic filter that easily extends its order is presented and discussed. 相似文献
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FIR数字滤波器具有稳定性和线性相位的特点,这对于要求高保真度的信号处理有很重要意义。利用MATLAB实现FIR低通数字滤波器的设计,并对被高频干扰的信号进行滤波,达到了预期结果。 相似文献
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A digital signal processing approach to interpolation 总被引:2,自引:0,他引:2
In many digital signal precessing systems, e.g., vacoders, modulation systems, and digital waveform coding systems, it is necessary to alter the sampling rate of a digital signal Thus it is of considerable interest to examine the problem of interpolation of bandlimited signals from the viewpoint of digital signal processing. A frequency dmnain interpretation of the interpolation process, through which it is clear that interpolation is fundamentally a linear filtering process, is presented, An examination of the relative merits of finite duration impulse response (FIR) and infinite duration impulse response (IIR) digital filters as interpolation filters indicates that FIR filters are generally to be preferred for interpolation. It is shown that linear interpolation and classical polynomial interpolation correspond to the use of the FIR interpolation filter. The use of classical interpolation methods in signal processing applications is illustrated by a discussion of FIR interpolation filters derived from the Lagrange interpolation formula. The limitations of these filters lead us to a consideration of optimum FIR filters for interpolation that can be designed using linear programming techniques. Examples are presented to illustrate the significant improvements that are obtained using the optimum filters. 相似文献
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We present an algorithmic approach to the design of low-power frequency-selective digital filters based on the concepts of adaptive filtering and approximate processing. The proposed approach uses a feedback mechanism in conjunction with well-known implementation structures for finite impulse response (FIR) and infinite impulse response (IIR) digital filters. Our algorithm is designed to reduce the total switched capacitance by dynamically varying the filter order based on signal statistics. A factor of 10 reduction in power consumption over fixed-order filters is demonstrated for the filtering of speech signals 相似文献
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This paper presents a method for the frequency domain design of infinite impulse response (IIR) digital filters. The proposed method designs filters approximating prescribed magnitude and phase responses. IIR filters of this kind can have approximately linear-phase responses in their passbands, or they can equalize magnitude and phase responses of given systems. In many cases, these filters can be implemented with less memory and with fewer computations per output sample than equivalent finite impulse response (FIR) digital filters. An important feature of the proposed method is the possibility to specify a maximum radius for the poles of the designed rational transfer function. Consequently, stability can be guaranteed, and undesired effects of implementations using fixed-point arithmetic can be alleviated by restricting the poles to keep a prescribed distance from the unit circle. This is achieved by applying Rouche's theorem in the proposed design algorithm. We motivate the use of IIR filters with an unequal number of poles and zeros outside the origin of the complex plane. In order to satisfy simultaneous specifications on magnitude and phase responses, it is advantageous to use IIR filters with only a few poles outside the origin of the z-plane and an arbitrary number of zeros. Filters of this type are a compromise between IIR filters with optimum magnitude responses and phase-approximating FIR filters. We use design examples to compare filters designed by the proposed method to those obtained by other methods. In addition, we compare the proposed general IIR filters with other popular more specialized structures such as FIR filters and cascaded systems consisting of frequency-selective IIR filters and phase-equalizing allpass filters 相似文献
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基于DSP的FIR滤波器的C语言算法实现 总被引:1,自引:0,他引:1
有限冲激响应(FIR)滤波器是数字信号处理系统中最基本的元件,具有严格的线性相频特性,同时其单位抽样响应是有限长的,系统稳定。阐述了FIR的基本原理,并进行了MATLAB仿真。基于TI公司的TMS320VC5402 DSP硬件平台,设计了FIR低通滤波器。采用C语言算法,利用集成开发环境代码调式器(Code Composer Studio,CCS)分别观察了输入和输出波形,验证了此算法的准确性和高效性,对信号处理及信号传输有重要的研究意义。 相似文献
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It is well known that IIR digital filters require quite fewer computations,comparedwith FIR filters,in order to meet stringent magnitude specifications when the phase distortioncan be tolerated.An approximately linear phase,however,can be also obtained with the IIRfilter by making use of a technique without increasing the complexity.Based on a certain numberof attenuation zeros in the pass band,a new approach is developed for the design of polyphasewave digital filters with exact magnitude responses and Chebyshev approximation of the desiredphase responses.The minimum number of attenuation zeros is estimated,and some examples areincluded. 相似文献
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在“数字信号处理”课程的学习和教学中,数字滤波器设计中采样频率的作用是一个较难理解和容易混淆的概念,本文详细讨论了采和频率在数字滤波器设计中的作用和影响,从数学原理和物理概念两方面说明了采样频率的作用以及产生概念错误的原因。 相似文献