FIR Photonic Microwave Filter Design Employing Simulated Annealing Algorithm

Photonic microwave filters are important parts of fiber‐optic microwave/millimeter wave processing systems. In this paper, the synthesis problem of fiber‐optic FIR microwave filters is addressed and a novel method for designing photonic microwave filter employing a simulated annealing (SA) algorithm is proposed. The design problem of photonic microwave filter can be reduced to be a multi‐variable function optimization problem, which can be solved by a simulated annealing‐based algorithm. As an illustration of the application of SA to photonic microwave filter design, the synthesis of an optimizedFBG‐based photonic microwave filter is discussed. Numerical results show that the global minimum finding capability of SA makes it be an efficient way to design the photonic microwave filter. Numerical results also demonstrate that the proposed method can be applied to design different filtering systems with different constraints.     

An approximation technique for the design of recursive digital and continuous filters with and without fixed zeros in the stop band

The bilinear z-transformation has been used by various authors (Golden and Kaiser 1964, Broomer 1966) to design recursive digital filters. The necessity to select an equivalent. continuous filter makes this method very laborious. Remes exchange algorithm has also been used in the design of recursive filters (Remes 1957, Deczky 1974) but not will the same success as for the design of non-recursive filters (McClennan and Parks 1973. Holt et al. 1975 ) due to its dependence on the initial conditions on optimization methods. In this paper we-present a design technique, based on Remes second exchange algorithm. This provides a versatile approximation method for the design of a wide range of minimax-type digital recursive filters. Furthermore, the method is extended to the design of continuous filters using suitable mapping methods. In the examples given, filters arc designed with various frequency characteristics for the pass band and the stop band and also with predefined zeros in the stop band.     

PCLS IIR digital filters with simultaneous frequency responsemagnitude and group delay specifications

This paper presents the peak-constrained least-squares (PCLS) approach to designing IIR digital filters. PCLS IIR digital filters that meet simultaneous specifications on the frequency response magnitude and the group delay are introduced. As a point of reference, we consider the IIR digital filter design problem that appears in Deczky's (1972) classic paper and in the popular textbook by Oppenheim and Schafer (1989). In addition, the same design problem appears in the IIR filter design chapter by Higgins and Munson (1993) in the Handbook for Digital Signal Processing. By using our new algorithm with simultaneous optimization of the frequency response magnitude and the group delay, we obtain a dramatic improvement in the solution of this classic IIR digital filter design problem. Starting from the same filter structure and the same specifications for the frequency response magnitude as in the works of Deczky, Oppenheim and Schafer, and Higgins and Munson, we are able to reduce the group delay ripple by a factor of 35. In another design problem that originated in Deczky's work, we use PCLS optimization to reduce the group delay ripple by a factor of 40 at the same time we reduce the stopband energy by 6 dB, without sacrificing any other performance measure. The group delay ripple in this IIR digital filter example is reduced to only ±0.002 samples     

New algorithm for fir filter design with discrete coefficients

This paper develops a new algorithm based on the Projected Gradient Algorithm (PGA) for the design of FIR digital filters with “sum of power of two” coefficients. It is shown that the integer programming involved in the FIR filter design can be solved by this algorithm.It is compared with the reported method for a SemiDefinite Programming (SDP) relaxationbased design. The simulations demonstrate that the new algorithm often yields the similar error performances of the FIR filter design, but the average CPU time of this approach is significantly reduced.     

巴特沃斯数字陷波滤波器的设计

随着数字技术的发展,数字滤波器在许多领域得到广泛的应用。研究一种在Matlab语言环境下设计IIR数字陷波滤波器的方法,在数字陷波滤波器设计过程中,先进行模拟低通滤波器的设计,然后进行模拟低通/模拟带阻滤波器转换,最后采用双线性变化法将模拟陷波滤波器转化成数字陷波滤波器。提出一种用所有零点和极点来表达数字陷波器传递函数的方法,同时给出以巴特沃斯模拟低通为原型设计数字陷波滤波器的程序。     

WCDMA系统中SPT数字滤波器的一种设计方法

相对于传统的数字滤波器实现方法,符号的2次幂和(SPT)滤波器用移位寄存器替代乘法器,因而资源消耗少、速度快,更加适于专用集成电路(ASIC)设计.介绍了一种适用于宽带码分多址(WCDMA)前向信道中给定基带成形滤波器单位脉冲响应后设计SPT滤波器的方法.相对于传统的理想SPT系数优化方法,此方法更适于给定单位脉冲响应后SPT滤波器的设计,计算复杂度低;仿真结果显示,相对于更加简单的四舍五入方法,此算法在性能上又有可观的增益.     

An iterative algorithm for optimal design of non-frequency-selective FIR digital filters

This paper proposes a novel iterative algorithm for optimal design of non-frequency-selective Finite Impulse Response （FIR） digital filters based on the windowing method. Different from the traditional optimization concept of adjusting the window or the filter order in the windowing design of an FIR digital filter, the key idea of the algorithm is minimizing the approximation error by successively modifying the design result through an iterative procedure under the condition of a fixed window length. In the iterative procedure, the known deviation of the designed frequency response in each iteration from the ideal frequency response is used as a reference for the next iteration. Because the approximation error can be specified variably, the algorithm is applicable for the design of FIR digital filters with different technical requirements in the frequency domain. A design example is employed to illustrate the efficiency of the algorithm.     

Particle Filter-Based Synchronization of Chaotic Colpitts Circuits Combating AWGN Channel Distortion

In this paper, synchronization of chaotic Colpitts circuits using a particle filter (PF) to combat the additive white Gaussian noise (AWGN) channel effect is studied by numeric simulations. A novel PF algorithm suitable for chaos synchronization is proposed. With this algorithm, chaos synchronization of Colpitts circuits can be achieved and maintained in AWGN channels. Parameters in the proposed PF algorithm are studied to understand their effects on synchronization performance. The synchronization performance using the proposed PF algorithm is compared with those using other digital filters, such as the extended Kalman filter and the generic PF. It is found that the proposed PF algorithm performs better than the other digital filters. Simulation results also show that the particle number is not very critical to the synchronization performance when this PF algorithm is used.     

Design Of 2-D Recursive Digital Filters Using Nonsymmetric Half-Plane Allpass Filters

A novel structure using recursive nonsymmetric half-plane (NSHP) digital allpass filters (DAFs) is presented for designing 2-D recursive digital filters. First, several important properties of 2-D recursive DAFs with NSHP support regions for filter coefficients are investigated. The stability of the 2-D recursive NSHP DAFs is guaranteed by using a spectral factorization-based algorithm. Then, the important characteristics regarding the proposed novel structure are discussed. The design problem of 2-D recursive digital filters using the novel structure is considered. We formulate the problem by forming an objective function consisting of the weighted sum of magnitude, group delay, and stability-related errors. A design technique using a trust-region Newton-conjugate gradient method in conjunction with the analytic derivatives of the objective function is presented to efficiently solve the resulting optimization problem. The novelty of the presented 2-D structure is that it possesses the advantage of better performance in designing a variety of 2-D recursive digital filters over existing 2-D filter structures. Finally, several design examples are provided for conducting illustration and comparison.     

A unified approach to the development of an exceedingly efficient algorithm for digital filter design

A design is presented for a digital filter, using the powerful alternation theorem and Remez exchange algorithm. The approach unifies all types of filters and the algorithm developed is exceedingly efficient as it is capable of designing a filter with 100-point impulse response in ca. 20 s.     

Digital Filter Design Using Computer Algebra Systems

Digital filter design can be performed very efficiently using modern computer tools. The drawback of the numeric-based tools is that they usually generate a tremendous amount of numeric data, and the user might easily lose insight into the phenomenon being investigated. The computer algebra systems successfully overcome some problems encountered in the traditional numeric-only approach. In this paper, we introduce an original approach to algorithm development and digital filter design using a computer algebra system. The main result of the paper is the development of an algorithm for an infinite impulse response (IIR) filter design that, theoretically, is impossible to be implemented using the traditional approach. We present a step-by-step procedure which includes derivations of closed-form expressions for (1) the transfer functions of the implemented digital filter which contains the algebraic loop; (2) the closed-form expression for computing the number of requested iteration steps; and (3) the error function representing the difference of the output sample values of the new filter and that of the conventional filter. We demonstrate how one can use some already-known multiplierless digital filter as a start-up filter to design a new digital filter whose passband edge frequency can be simply moved by using a single parameter. As a result, we obtain a multiplierless IIR filter, which belongs to the family of low-power digital filters where multipliers are replaced with a small number of adders and shifters.     

Digital filter design techniques in the frequency domain

Digital filtering is the process of spectrum shaping using digital components as the basic elements. Increasing speed and decreasing size and cost of digital components make it likely that digital filtering, already used extensively in the computer simulation of analog filters, will perform, in real-time devices, the functions which are now performed almost exclusively by analog components. In this paper, using the z-transform calculus, several digital filter design techniques are reviewed, and new ones are presented. One technique can be used to design a digital filter whose impulse response is like that of a given analog filter; other techniques are suitable for the design of a digital filter meeting frequency response criteria. Another technique yields digital filters with linear phase, specified frequency response, and controlled impulse response duration. The effect of digital arithmetic on the behavior of digital filters is also considered.     

Design of hybrid filter banks for analog/digital conversion

This paper presents design algorithms for hybrid filter banks (HFBs) for high-speed, high-resolution conversion between analog and digital signals. The HFB is an unconventional class of filter bank that employs both analog and digital filters. When used in conjunction with an array of slower speed converters, the HFB improves the speed and resolution of the conversion compared with the standard time-interleaved array conversion technique. The analog and digital filters in the HFB must be designed so that they adequately isolate the channels and do not introduce reconstruction errors that limit the resolution of the system. To design continuous-time analog filters for HFBs, a discrete-time-to-continuous-time (“Z-to-S”) transform is developed to convert a perfect reconstruction (PR) discrete-time filter bank into a near-PR HFB; a computationally efficient algorithm based on the fast Fourier transform (FFT) is developed to design the digital filters for HFBs. A two-channel HFB is designed with sixth-order continuous-time analog filters and length 64 FIR digital filters that yield -86 dB average aliasing error. To design discrete-time analog filters (e.g., switched-capacitors or charge-coupled devices) for HFBs, a lossless factorization of a PR discrete-time filter bank is used so that the reconstruction error is not affected by filter coefficient quantization. A gain normalization technique is developed to maximize the dynamic range in the finite-precision implementation. A four-channel HFB is designed with 9-bit (integer) filter coefficients. With internal precision limited to the equivalent of 15 bits, the maximum aliasing error is -70 dB, and with the equivalent of 20 bits internal precision, maximum aliasing is -100 dB. The 9-bit filter coefficients degrade the stopband attenuation (compared with unquantized coefficients) by less than 3 dB     

Design of High-Order Extrapolated Impulse Response FIR Filters with Signed Powers-of-Two Coefficients

Expensive multiplication operations can be replaced by simpler additions and hardwired shifters so as to reduce power consumption and area size, if the coefficients of a digital filter are signed power-of-two (SPT). As a consequence, FIR digital filters with SPT coefficients have been widely studied in the last three decades. However, most approaches for the design of FIR filters with SPT coefficients focus on filters with length less than 100. These approaches are not suitable for the design of high-order filters because they require excessive computation time. In this paper, an approach for the design of high-order filters with SPT coefficients is proposed. It is a two-step approach. Firstly, the design of an extrapolated impulse response (EIR) filter is formulated as a standard second-order cone programming (SOCP) problem with an additional coefficient sensitivity constraint for optimizing its finite word-length effect. Secondly, the obtained continuous coefficients are quantized into SPT coefficients by recasting the filter-design problem into a weighted least squares (WLS) sequential quadratic programming relaxation (SQPR) problem. To further reduce implementation complexity, a graph-based common subexpression elimination (CSE) algorithm is utilized to extract common subexpressions between SPT coefficients. Simulation results show that the proposed method can effectively and efficiently design high-order SPT filters, including Hilbert transformers and half-band filters with SPT coefficients. Experiment results indicate that 0.81N∼0.29N adders are required for 18-bit N-order FIR filters (N=335∼3261) to meet the given magnitude response specifications.     

A fast algorithm for FIR digital filtering with a sum-of-triangles weighting function

A doubly recursive algorithm for time domain convolution with a piecewise linear weighting function is presented that combines the speed of a recursive (IIR) digital filter with the flexibility and ease of design of a nonrecursive (FIR) digital filter. The approach approximates the desired FIR weighting function by a sum-of-triangles weighting function. ForL triangles (or triangle pairs for a linear phase filter) the algorithm is of orderLN. The approximation improves with the number of triangles. A significant advantage of the algorithm compared to FFT filtering or direct convolution is that there is no necessity of a tradeoff between frequency response accuracy and computation time per output point as the data spacing decreases in the filtered signal. The computational complexity is dependent on the number of triangles chosen, not the width of the weighting function, so the algorithm is especially effective for filters with an inherently wide FIR weighting function.     

Design of arbitrary-phase variable digital filters using SVD-based vector-array decomposition

This paper proposes a straightforward method for designing variable digital filters with arbitrary variable magnitude as well as arbitrary fixed-phase or variable fractional delay (VFD) responses. The basic idea is to avoid the complicated direct design of one-dimensional (1-D) variable digital filters by decomposing the original variable filter design problem into easier subproblems that only require constant 1-D filter designs and multidimensional polynomial approximations. Through constant 1-D filter designs and multidimensional polynomial fits, we can easily obtain a variable digital filter satisfying the given variable design specifications. To decompose the original variable filter design into constant 1-D filter designs and multidimensional polynomial fits, a new multidimensional complex array decomposition called vector array decomposition (VAD) is proposed, which is based on two new theorems using the singular value decomposition (SVD). Once the VAD is obtained, the subproblems can be easily solved. Furthermore, we show that the VAD can also be generalized to the weighted least squares (WLS) case (WLS-VAD) for the WLS variable filter design. Three design examples are given to illustrate that the WLS-VAD and VAD-based techniques are considerably efficient for designing variable digital filters with arbitrary variable magnitude and arbitrary fixed-phase or VFD responses.     

Complex coefficient fir digital filters

The complex FIR digital filter is a filter that has complex coefficients in itsZ-domain transfer function. The set of coefficients is determined, based on some criterion, to meet predefined requirements. On this basis, an algorithm is proposed for designing FIR digital filters with asymmetric amplitude response in conjunction with linear phase. Minimax approximation has been adopted for determining the set of coefficients, where the associated set of overdetermined linear equations is solved by using an efficient linear programming algorithm. Computer simulations show that, to meet prescribed specifications, the proposed design algorithm yields a complex FIR digital filter with the lowest order.     

基于FPGA的FIR滤波器设计

数字滤波器在数字信号处理中占有很重要的地位,该文介绍了FIR滤波器的两种实现算法：乘累加算法和优化的分布式算法,其中分布式算法作为优化算法进行研究。其次,根据FIR滤波器理论,采用线性相位结构优化滤波器的设计。并给出了FIR滤波器的模块划分和FIR滤波器的主要模块的实现,最后对FIR滤波器进行了系统仿真和验证。     

Efficient design of digital filters for 2-D and 3-D depth migration

Two- and three-dimensional (2-D and 3-D) depth migration can be performed using 1-D and 2-D extrapolation digital filters, respectively. The depth extrapolation is done, one frequency at a time, by convolving the seismic wavefield with a complex-valued, frequency- and velocity-dependent, digital filter. This process requires the design of a complete set of extrapolation filters: one filter for each possible frequency-velocity pair. Instead of independently designing the frequency- and velocity-dependent filters, an efficient procedure is introduced for designing a complete set of 1-D and 2-D extrapolation filters using transformations. The problem of designing a desired set of migration filters is thus reduced to the design of a single 1-D filter, which is then mapped to produce all the desired 1-D or 2-D migration filters. The new design procedure has the additional advantage that both the 1-D and 2-D migration filters can be realized efficiently and need not have their coefficients precomputed or tabulated     

A simple and efficient design of variable fractional delay FIR filters

In this paper, a simple and efficient approach for designing one-dimensional variable fractional delay finite impulse response digital filters is proposed. Two matrix equations, based respectively on the weighted least-squares function of the optimum fixed fractional delay filter and the filter coefficient polynomial fitting, are formulated in tandem to form the design algorithm, which only has the computation complexity comparable with that of designing fixed finite impulse response digital filters. A design example is also given to justify the effectiveness and advantages of the proposed design method.