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。     

Fast LMS/Newton algorithms based on autoregressive modeling andtheir application to acoustic echo cancellation

We propose two new implementations of the LMS/Newton algorithm for efficient realization of long adaptive filters. We assume that the input sequence to the adaptive filter can be modeled as an autoregressive (AR) process whose order may be kept much lower than the adaptive filter length. The two algorithms differ in their structural complexity. The first algorithm, which will be an exact implementation of the LMS/Newton algorithm if the AR modeling assumption is accurate, is structurally complicated and fits best into a digital signal processing (DSP)-based implementation. On the other hand, the second algorithm is structurally simple and is tailored more toward very large-scale integrated (VLSI) custom chip design. Analyses of the proposed algorithms are given. It is found that for long filters, both algorithms perform about the same. However for short filters, a noticeable difference between the two may be observed. Simulation results that confirm our theoretical findings are given. Moreover, experiments with speech signals for modeling the acoustics of an office room show the superior convergence of the proposed algorithms when compared with the normalized LMS algorithm     

一种扬声器频响的自适应数字滤波优化补偿方法

从模拟滤波器到数字滤波器的转换以及寻找滤波器参数的最优解是IIR参数滤波器用于扬声器频响实时补偿的两个关键问题。本文提出一种根据模拟滤波器的中心频率、增益和品质因数来自适应选择S域到Z域转换方法,通过扬声器频响曲线的实时包络来划分峰谷,给出了一种滤波器参数的优化方法,提升了数字滤波器与模拟原型的匹配度,从而减少滤波器个数、加速了滤波器的优化计算。实验结果表明,所提出的方法使补偿滤波器的个数和总时间开销比现有的补偿方法分别减少了大约43%和55%。     

Subspace approach for the design of cosine-modulated filter bankswith linear-phase prototype filter

A method for designing perfect reconstruction (PR) prototypes for paraunitary cosine-modulated filter banks is presented. The design procedure is based on a subspace approach that allows linear combinations of even-length linear-phase PR prototype filters in such a way that the resulting filter is also a linear-phase PR prototype. Within a given subspace, the weights of the optimal linear combination can easily be computed via an eigenanalysis. The filter design is carried out iteratively while the PR property is guaranteed throughout the design process. No nonlinear optimization routine is needed. As a special case, the proposed approach allows the design of discrete-coefficient prototypes, which are of great interest for efficient hardware implementations     

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.     

Adaptive inverse filters for stereophonic sound reproduction

A general theoretical basis for the design of adaptive digital filters used for the equalization of the response of multichannel sound reproduction systems is described. The approach is applied to the two-channel case and then extended to deal with arbitrary numbers of channels. The intention is to equalize not only the response of the loudspeakers and the listening room but also the crosstalk transmission from right loudspeaker to left ear and vice versa. The formulation is a generalization of the Atal-Schroeder crosstalk canceler. However, the use of a least-squares approach to the digital filter design and of appropriate modeling delays potentially allows the effective equalization of nonminimum phase components in the transmission path. A stochastic gradient algorithm which facilitates the adaptation of the digital filters to the optimal solution, thereby providing the possibility of designing the filters in situ, is presented. Some experimental results for the two-channel case are given     

A custom digital intermediate frequency filter for the Americanmobile telephone system

A digital filter for intermediate frequency filtering in mobile communication systems is presented. The purpose of the work is to show an alternative to the analog filters which are used in most of today's heterodyne receivers. Bit-serial arithmetic is applied on a twelfth-order wave digital lattice filter algorithm. The paper also shows a method for retiming such algorithms. The power consumption in two fabricated prototypes is compared. By customizing the library cells, the power consumption has been reduced significantly. In the low power prototype, the power dissipation is 8 mW using 3 V supply voltage. The prototype is a 10 MIPS design fabricated in a 0.8-μm standard two-metal-layer CMOS process     

QPSO算法在FIR数字滤波器设计中的应用

FIR数字滤波器的设计实际上是一个多维变量寻优问题,滤波器的设计可转化为滤波器参数优化的问题。文章介绍了粒子群优化算法、量子粒子群优化算法,然后利用算法对参数空间进行搜索以获得参数的最优化,根据预期频率特性的设计要求,通过仿真实验表明设计方法的有效性。     

Adaptive reduced-rank LCMV beamforming algorithms based on joint iterative optimization of filters: Design and analysis

This paper presents reduced-rank linearly constrained minimum variance (LCMV) beamforming algorithms based on joint iterative optimization of filters. The proposed reduced-rank scheme is based on a constrained joint iterative optimization of filters according to the minimum variance criterion. The proposed optimization procedure adjusts the parameters of a projection matrix and an adaptive reduced-rank filter that operates at the output of the bank of filters. We describe LCMV expressions for the design of the projection matrix and the reduced-rank filter. We then describe stochastic gradient and develop recursive least-squares adaptive algorithms for their efficient implementation along with automatic rank selection techniques. An analysis of the stability and the convergence properties of the proposed algorithms is presented and semi-analytical expressions are derived for predicting their mean squared error (MSE) performance. Simulations for a beamforming application show that the proposed scheme and algorithms outperform in convergence and tracking the existing full-rank and reduced-rank algorithms while requiring comparable complexity.     

Realisation of a switched-capacitor voltage wave pseudobandpass filter

This letter presents a novel approach to the realisation of monolithic pseudobandpass filters using NMOS technology. The method is based on the use of sampled analogue signals and is related to the wave digital filter in its design techniques. Results are presented for a prototype pseudobandpass filter which uses the periodicity of a cascaded transmission line microwave filter.     

Modified Metaheuristic Algorithms for the Optimal Design of Multiplier-Less Non-uniform Channel Filters

Reconfigurable non-uniform channel filters are now being widely used in software define radio (SDR). The hardware implementation of these filters requires low complexity, low chip area and low power consumption. The frequency response masking (FRM) approach is proved to be a good candidate for the realization of a sharp digital finite impulse response (FIR) filter with low complexity. To reduce the complexity further, this paper gives an optimal design method which makes the channel filters totally multiplier-less. This is done in two steps. The channel filters are designed using the FRM approach with continuous filter coefficients. To obtain multiplier-less design, these filter coefficients are converted to finite-precision coefficients using signed power of two (SPT) space and the filter coefficients are synthesized in the canonic signed-digit (CSD) format. But this may lead to degradation of the filter performance. Hence the filter coefficients synthesis in the CSD format is formulated as an optimization problem. Several meta-heuristic algorithms like Differential Evolution (DE), Artificial Bee Colony (ABC), Harmony Search Algorithm (HSA) and Gravitational Search Algorithm (GSA) are modified and deployed and the best one is selected.     

Low-Area/Power Parallel FIR Digital Filter Implementations

This paper presents a novel approach for implementing area-efficient parallel (block) finite impulse response (FIR) filters that require less hardware than traditional block FIR filter implementations. Parallel processing is a powerful technique because it can be used to increase the throughput of a FIR filter or reduce the power consumption of a FIR filter. However, a traditional block filter implementation causes a linear increase in the hardware cost (area) by a factor of L, the block size. In many design situations, this large hardware penalty cannot be tolerated. Therefore, it is important to design parallel FIR filter structures that require less area than traditional block FIR filtering structures. In this paper, we propose a method to design parallel FIR filter structures that require a less-than-linear increase in the hardware cost. A novel adjacent coefficient sharing based sub-structure sharing technique is introduced and used to reduce the hardware cost of parallel FIR filters. A novel coefficient quantization technique, referred to as a scalable maximum absolute difference (MAD) quantization process, is introduced and used to produce quantized filters with good spectrum characteristics. By using a combination of fast FIR filtering algorithms, a novel coefficient quantization process and area reduction techniques, we show that parallel FIR filters can be implemented with up to a 45% reduction in hardware compared to traditional parallel FIR filters.     

Optimal design of transform-based block digital filters using a quadratic criterion

Block digital filtering is a powerful tool to reduce the computational complexity of digital filtering systems. However, due to their block structure, block digital filters (BDFs) are time-varying linear systems, hence, their design is not easy. The most widely spread approaches to BDF design consist of constraining the BDF to be time-invariant (by restricting the design process to a specific subset of possible solutions) and then using conventional filter synthesis techniques. In this paper, we do not restrict the design process, and we propose a simple and optimal matrix-oriented approach to optimize the BDF coefficients. Furthermore, the proposed approach takes profit of the structure of transform-based BDFs to considerably reduce the computational complexity and memory requirements of the design process. Experimental results confirm that as expected, the obtained global distortion is lower than the distortion obtained with a traditional technique such as overlap-save.     

A generalized parametric PR-QMF design technique based on Bernsteinpolynomial approximation

A generalized, parametric, perfect-reconstruction quadrature-mirror-filter (PR-QMF) design technique based on Bernstein polynomial approximation in the magnitude-square domain is presented. The parametric nature of this solution provides useful insights to the PR-QMF problem. Several well-known orthonormal wavelet filters, PR-QMFs, are shown to be the special cases of the proposed technique. Energy compaction performances of a few popular signal decomposition techniques are presented for AR(1) signal sources. It is observed that the hierarchical QMF filter banks considered outperform the block transforms as expected     

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.     

Design of minimum-phase digital filters as the sum of two allpass functions using the cepstrum technique

A new and practical approach using the cepstrum technique is proposed in the design of minimum-phase digital filters as the sum of two allpass functions. The desired magnitude response is specified in the frequency domain. Its corresponding minimum-phase response is then obtained from the desired magnitude response. The desired phases for the two allpass filters are obtained from the magnitude and phase responses. For both filters to be stable, the corresponding denominator polynomials are minimum phase. The filter coefficients are obtained from the desired phases using the cepstrum technique. Design examples show that the method works well for both classical filter specification and general magnitude specification in the frequency domain.     

基于FIR与相位补偿IIR滤波器的雷达回波信号处理

FIR与IIR频率选择滤波器的设计,被广泛应用于数字信号处理领域之中。文章以雷达回波信号的数字处理为例,首先分别设计FIR,IIR滤波器完成了对信号特定频率分量的滤除。进而,针对IIR滤波器的非线性相位,基于最优化设计全通系统实现了相位补偿,并对FIR,IIR滤波器进行了综合比较。     

Design techniques for high-speed multirate multistage FIR digital filters

This paper presents architecture design techniques for implementing both single-rate and multirate high-speed finite impulse response (FIR) digital filters, with emphasis on the multirate multistage interpolated FIR (IFIR) digital filters. Well-known techniques to achieve high-speed and low-power applications for the single-rate digital FIR architecture are summarized, followed by the introduction of variable filter order selection, optimal filter decomposition, memory-saving and mirror symmetric filter pairs techniques which offer further gains in both performance and complexity reduction for the multirate multistage digital FIR architecture. A filter design example with TSMC 0.25?µm standard cell for 64-QAM baseband demodulator shows that the area is reduced by 39% for low-complexity application. Moreover, for high-speed application, the chip can operate at 714?MHz. Finally, a designed decimator which is used in the CDMA cellular shows that the area is reduced by 70% as compared with conventional approach.     

An Efficient Algorithm for the Optimization of FIR Filters Synthesized Using the Multistage Frequency-Response Masking Approach

A very efficient technique to drastically reduce the number of multipliers and adders in narrow transition-band linear-phase finite-impulse response digital filters is to use the one-stage or multistage frequency-response masking (FRM) approach, which has been originally introduced by Lim and further improved by Lim and Lian. In these original synthesis techniques, the subfilters in the overall implementation are separately designed. As shown earlier by the authors of this contribution together with Johansson, the arithmetic complexity in one-stage FRM filter designs can be considerably reduced by using the following two-step technique for simultaneously optimizing all the subfilters. First, a suboptimal solution is found by using a simple design scheme. Second, this solution is used as a start-up solution for further optimization, which is carried out with the aid of an efficient nonlinear optimization algorithm. This paper exploits this approach to synthesizing multistage FRM filters. An example taken from the literature illustrates that both the number of multipliers and the number of adders for the resulting optimized multistage FRM filters are approximately 70 percent compared with those of the filters synthesized using the original multistage FRM filter design schemes. Additional examples are included in order to show the benefits provided by the proposed synthesis scheme over other recently published design techniques, in terms of an improved performance of the resulting solution, a higher accuracy of the solution, and a faster speed required to arrive at the best solution.     

Pareto Optimal Microwave Filter Design Using Multiobjective Differential Evolution

Microwave filters play an important role in modern wireless communications. A novel method for the design of multilayer dielectric and open loop ring resonator (OLRR) filters under constraints is presented. The proposed design method is based on generalized differential evolution (GDE3), which is a multiobjective extension of differential evolution (DE). GDE3 algorithm can be applied for global optimization to any engineering problem with an arbitrary number of objective and constraint functions. GDE3 is compared against other evolutionary multiobjective algorithms like nondominated sorting genetic algorithm-II (NSGA-II), multiobjective particle swarm optimization (MOPSO) and multiobjective particle swarm optimization with fitness sharing (MOPSO-fs) for a number of microwave filter design cases. In the multilayer dielectric filter design case a predefined database of low loss dielectric materials is used. The results indicate the advantages of this approach and the applicability of this design method.