Low-complexity linear phase fir filters in cascade form

Digital filters play a central role in modern Digital Signal Processing (DSP) systems. Finite Impulse Response (FIR) filters can provide solutions with guaranteed stability and linear phase. However, the main disadvantage of conventional FIR filter designs is that they become computationally complex, especially in applications demanding narrow transition bandwidths. Therefore, designing FIR filters with very stringent specifications and a low complexity is currently an important challenge. In this chapter, a review of the recent methods to efficiently design low-complexity linear-phase FIR filters is presented. The chapter starts with an introduction to linear-phase FIR digital filters. Then, an overview of the design methods that have been developed in literature to design low-complexity FIR filters is presented. Finally, the most common and recent of these methods along with their corresponding special structures are explained.

Digital filters play a central role in modern digital signal processing (DSP) systems. Finite impulse response (FIR) filters can provide solutions with guaranteed stability and linear phase. However, the main disadvantage of conventional FIR filter designs is that they become computationally complex, especially in applications demanding narrow transition bandwidths. Therefore, designing FIR filters with very stringent specifications and a low complexity is currently an important challenge. In this chapter, a review of the recent methods to efficiently design low-complexity linear-phase FIR filters is presented. The chapter starts with an introduction to linear-phase FIR digital filters. Then, an overview of the design methods that have been developed in literature to design low-complexity FIR filters is presented. Finally, the most common and recent of these methods along with their corresponding special structures are explained.

Digital filters in cascade form enjoy many advantages over their equivalent single-stage realizations in that lower coefficient sensitivity, higher throughput, reduced computational and smaller implementation cost can be achieved. However, the numerical design and optimization of such structure are of much more difficulty than the single-stage case if the filter coefficients are restricted to be of discrete values. This is mainly due to the non-convexity of the constraints, which rules out the possibility of employing sophisticated convex optimization techniques as well as the guaranteed global optimality. In this work, a general-purpose algorithm is proposed for the design of linear phase finite impulse response (FIR) filters in cascade form with discrete coefficients. The proposed algorithm decomposes the overall filter into subfilters during the traverse of a tree search of the overall filter. Discrete-valued linear phase FIR filters are able to be searched and decomposed into both symmetric and non-symmetric subfilters. The optimization complexity is of the same order as the single-stage filter optimization. Design examples have shown that the proposed algorithm is capable of achieving notable reduction in both implementation cost and adder depth compared with their single-stage optimum designs.

A novel procedure that makes use of the maximum entropy method (MEM) for the design of linear-phase FIR (finite impulse response) digital filters is described. It is shown that, by applying MEM to the inverse of the desired gain function or its square root function, it is possible to generate linear-phase FIR filters that match the given gain function to any desired degree of accuracy. Moreover, an iterative algorithm makes the design procedure very efficient, since the higher-order filters can be recursively generated from the lower-order ones. To minimize the effect of any residual passband distortion, a final averaging scheme on the lower-order filters generates a class of passband distortion-free linear-phase FIR filters. The new filters are seen to be more robust to changes in transition bandwidth as well as stopband level. Simulation results that compare the present procedure with other well-known methods are also presented.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

This correspondence presents a novel approach for the design of two channel perfect reconstruction (PR) linear phase finite-impulse response (FIR) filter banks. A broad class of even and odd filter length PR filter bank cases can be obtained by simply reordering an appropriate cascade form synthesizing the polyphase components of a linear phase FIR filter.

The weighted least square (WLS) design of two-dimensional (2-D) linear-phase finite impulse response (FIR) filters with quadrantally symmetric or anti-symmetric magnitude responses is studied in this paper. Firstly, the necessary and sufficient condition for minimizing the WLS error is obtained in the form of matrix equation. Based on the matrix equation, a novel algorithm is derived for the WLS design of 2-D filters with two-valued weighting functions. Further, convergence of the algorithm is established and the computational complexity is analyzed. Because the new algorithm deals with the filter parameters in their natural 2-D form and the transition band is not sampled, the amount of computation required is reduced significantly, especially for high order cases. Finally, examples are presented to illustrate the performance of the proposed algorithm.

Recursive structures are introduced for implementing long linear-phase finite impulse response (FIR) filters using a very small number of multipliers. The implementation uses the principle of switching and resetting between two identical copies of the same infinite impulse response (IIR) filter. The impulse response of the resulting filters is a truncated and shifted version of the response of a filter G(z)G(z/sup -1/) where G(z) is a stable IIR stable and G(z/sup -1/) is the corresponding unstable IIR filter. The filters are implemented as a parallel connection of several branches, each branch generating a truncated response corresponding to a complex conjugate pole pair and its mirror-image pair. The truncation is performed using a feedforward term which provides pole-zero cancellations. To stabilize the pole-zero cancellations, and to avoid the quantization error from growing too much, the branch filters are implemented by applying the principle of switching and resetting. It is shown, by means of an example, that by using this approach a nearly optimum FIR filter of order larger than 500 using just 17 adjustable parameters can be designed. >

PurposeTraditionally, each time a new design for a finite impulse response (FIR) filter is required, a new algorithm has to be developed specially for the FIR filter. Furthermore, corresponding hardware architecture must be designed specially to meet the requirement of the FIR specifications. The purpose of this paper is to propose an arithmetic logic unit (ALU)‐based universal FIR filter suitable for realization in field programmable gate arrays (FPGA), where various FIR filters can be implemented just by programming instructions in the ROM with identical hardware architecture.Design/methodology/approachRather than multiplier‐accumulator‐based architecture for conventional FIR, the proposed ALU architecture implements the FIR functions by using accumulators and shift‐registers controlled by the instructions of ROM. Furthermore, time division multiplexing access (TDMA) technique is employed to reduce the chip size. In addition, the proposed FIR architecture is verified in a SOC hardware and/or software co‐emulation system.FindingsAn ALU‐based universal FIR filter suitable for realization in FPGA is designed and verified in a SOC hardware/software co‐emulation system with example of a 64‐tap FIR filter design.Originality/valueA software‐based design method as well as TDMA scheme for the ALU‐based FIR filter are introduced, making FIR filter architecture universal, programmable, and consuming less FPGA resources.

Chapter 2 - Design and Implementation of Digital FIR Filters

WITHDRAWN: An Optimistic Design of 16-Tap FIR Filter with Radix-4 Booth Multiplier Using Improved Booth Recoding Algorithm

Background/Objectives: FIR filters play a vital role in signal processing applications. This research work presents a low power and high speed efficient buffer based Distributed Algorithms and it is analyzed with Electro Cardio Gram (ECG) signal Finite Impulse Response (FIR) filter design. Methods/Statistical Analysis: The proposed FIR filter is designed using buffer based DA and it is simulated and synthesized using Cadence digital labs. This is compared with different architectures such as conventional DA, separated look up table DA and LUT less DA. Findings: Synthesis report shows that the proposed design has 52% less power dissipation compared with the conventional DA, 21% reduction in delay with separated Look Up Table (LUT) DA and 8% reduction in area with LUT less DA. Conclusion/Improvements: Presently this method is applied for ECG signal input with 16-tap FIR filter. This can be extended for higher order filters to achieve better performance.

In recent years, several approaches have been investigated to obtain multiplier-free realizations of digital filters. One approach makes use of periodically time-varying (PTV) structures. The idea is to distribute the computation in time and in space. Distribution in time provides reuse of the same hardware by means of PTV coefficients. Distribution in space increases the number of coefficients but simplifies the values of the coefficients. Computation distribution is based on radix-r number representation, and it can be carried out to the extent that each computation involves a simple coefficient that can be realized using only addition and shift (no hardware multiplier). Previous PTV realizations could not exploit the coefficient symmetry of finite impulse response (FIR) filters to reduce the number of coefficients. This paper proposes design and realizations of FIR filters with punctured radix-8 coefficients belonging to the septuple set {0, ±1, ±2, ±4), which can be implemented using only a shift operation without requiring any hardware multiplier. The realizations exploit the coefficient symmetry to reduce the hardware by about one-half. Due to a non-uniform grid of representation, we apply a modified Karmarkar's linear programming algorithm to find the optimum set of discrete coefficients that minimizes the weighted peak ripple error. Comparison with a conventional FIR filter with sum-of-powers-of-two (SOPOT) coefficients shows that the proposed filter is faster and uses less hardware than one with SOPOT coefficients. However, the punctured radix-8 system has a limit on the achievable ripple.

The Cook-Toom algorithm is widely used in short-length linear convolution, which is the building block of large points convolution algorithms. This paper proposes improved parallel finite impulse response (FIR) filter structures for linear-phase FIR filter, which is based on the Cook-Toom algorithm. In the proposed structures, Cook-Toom algorithm is used to reduce the number of sub-filters, and the symmetric properties of the linear-phase FIR filter's coefficients is used to further reduce the number of multipliers in sub-filters. Compared with the reported FFA and ISCA parallel FIR filter structures, the proposed method can substantially reduce the computational complexity. Specifically, for a 8-parallel 144-tap filter, the proposed design saves 18 multipliers (5%), 45 adders (7.9%) compared with the structure based on Winograd convolution algorithm [7].

This paper considers the design of computationally efficient narrow-band linear-phase finite-impulse response (FIR) filters with the transfer function of the form H(z) = F(z/sup M/)G(z), where F(z) is a linear-phase FIR transfer function. G(z) is constructed as a cascade of two linear-phase multiplier-free FIR filter building blocks. The first blocks are of the form 2/sup -P//sub r/[1-z/sup -K//sub r//(1-z/sup -1/) with different values of the integers K/sub r/. In these blocks the P/sub r/'s are integers. The second blocks are of the form ( 1/2 )(1 + z/sup -L//sub r/) with different values of the integer L/sub r/. Given the filter specifications, the problem is to find these building blocks, the order of F(z), and the integer M in F(z/sup M/) and to optimize the impulse-response values of F(z) so that the arithmetic complexity is minimized. It is shown how F(z) can be conveniently optimized with the aid of the fast Remez multiple exchange algorithm. For multiplier-free designs, the overall filter should exceed the given criteria in order to enable one to quantize the coefficient values of F(z/sup L/) to have two powers-of-two representations. Two examples taken from the literature show that the optimized proposed filters outperform other existing filter designs meeting the same criteria.

In this paper, the design of constrained linear phase finite impulse response (FIR) filters is considered. The problem is formulated as a linear complementarity problem (LCP), which is solved using Lemke's algorithm. The LCP is a refined mathematical formalism with useful theoretical results. The digital filters presented meet efficiently the specifications of the magnitude response error. The used algorithm is a direct one and therefore, there is no need for matrix inversion. However, in the iterative methods that are frequently used, the bulk of the design computation is concerned with the evaluation of matrix inversion in order to solve a system of equations. Examples to illustrate the proposed method are presented. Copyright © 2005 John Wiley & Sons, Ltd.