A Low Complexity Reconfigurable Multi-stage Channel Filter Architecture for Resource-Constrained Software Radio Handsets

Software defined radio (SDR) is emerging as a powerful platform for future generation cellular systems, due to its capability to operate conforming to multiple mobile radio standards. Channelizer in an SDR operates at the highest sampling rate and hence a low complexity design is needed for the most computationally intensive part of the SDR receiver. The channel filters in the channelizer extracts radio channels of varying bandwidths, corresponding to various communication standards from the wideband input signal. An architecture for implementing low complexity, low power and reconfigurable channel filter for the SDR mobile handsets, based on multi-stage frequency response masking (FRM) is proposed in this paper. The proposed architecture is unique in a way that it is able to effectively exploit the redundancy in multi-stage realization by utilizing the common masking filters and also capable of extracting varying bandwidth channels. Design examples show that the proposed architecture offers 47.5% complexity reduction and 18.1% power reduction over single-stage FRM approach.     

A Novel Diversity-Controlled Genetic Algorithm for Rapid Optimization of Bandpass FRM FIR Digital Filters Over CSD Multiplier Coefficient Space

The conventional frequency response masking (FRM) approach is one of the most well-known techniques for the design of sharp transition band finite impulse response (FIR) digital filters. The resulting FRM digital filters permit efficient hardware implementations due to an inherently large number of zero-valued multiplier coefficients. The hardware complexity of these digital filters can further be reduced by representing the remaining (non-zero) multiplier coefficient values by using their canonical signed-digit (CSD) representations. This paper presents a novel diversity-controlled (DC) genetic algorithm (GA) for the discrete optimization of bandpass FRM FIR digital filters over the CSD multiplier coefficient space. The resulting bandpass FIR digital filters are permitted to have equal or unequal lower and upper transition bandwidths. The proposed DCGA is based on an indexed look-up table of permissible CSD multiplier coefficients such that their indices form a closed set under the genetic operations of crossover and mutation. The salient advantage of DCGA over the conventional GA lies in the external control over population diversity and parent selection, giving rise to a rapid convergence to an optimal solution. The external control is achieved through the judicious choice of a pair of DCGA optimization parameters. An empirical investigation is undertaken for choosing appropriate values for these control parameters. The convergence speed advantages of the DCGA are demonstrated through its application to the design and optimization of a pair of bandpass FRM FIR digital filters with equal or arbitrary lower and upper transition bandwidths. In both cases, an increase of about an order of magnitude in the speed of convergence is achieved as compared to the conventional GAs.     

Reconfigurable Frequency Response Masking Filters for Software Radio Channelization

Low complexity and reconfigurability are two key requirements of channel filters in a software defined radio receiver. A new reconfigurable architecture based on frequency response masking (FRM) technique for the implementation of channel filters is proposed in this paper. Our architecture offers reconfigurability at filter and architecture levels, in addition to the inherent low complexity offered by the FRM technique. The proposed reconfigurable filter has been synthesized on 0.18- CMOS technology and implemented and tested on Virtex-II 2v3000ff1152-4 field-programmable gate array. Synthesis results show that the proposed channel filter offers average area and power reductions of 53.6% and 57.6%, respectively ,with average improvement in speed of 47.6% compared to other reconfigurable filters in literature.     

Design of non-uniform filter bank transmultiplexer with canonical signed digit filter coefficients

Design of non-uniform filter bank transmultiplexer (NUFB TMUX) with canonical signed digit (CSD) coefficients is presented. NUFB TMUX is preferred in a multicarrier communication system when applications with different data rates are to be multiplexed. If the filter coefficients are represented in CSD format, the hardware complexity of the NUFB TMUX can be reduced. A continuous coefficient NUFB TMUX is designed and the coefficients of the filters are synthesised in CSD format using genetic algorithm (GA). Separate objective functions are formulated for the fitness evaluation of the filters. Chromosomes are encoded as ternary digit strings. New crossover and mutation techniques are introduced to preserve the canonical property of the signed power of two (SPT) representations. For the fast convergence of the GA, positiondependent probability of mutation is used. Simulation results show that the CSD coefficient NUFB TMUX designed using the proposed algorithm has better signal-to-interference ratio (SIR) than that of continuous coefficient NUFB TMUX and CSD coefficient NUFB TMUX obtained by rounding. Frequency responses of its filters are better than that of the filters in CSD coefficient NUFB TMUX obtained by rounding and comparable with that of continuous coefficient NUFB TMUX.     

Two Classes of Frequency-Response Masking Linear-Phase FIR Filters for Interpolation and Decimation

This paper introduces two classes of frequency-response masking (FRM) linear-phase finite (length) impulse response (FIR) filters for interpolation and decimation by arbitrary integer factors M. As they are based on the FRM approach, the proposed filters are low-complexity (efficient) sharp-transition linear-phase FIR interpolation and decimation filters. Compared to previously existing FRM linear-phase FIR filter classes for interpolation and decimation, the new ones offer lower complexity and more freedom in selecting the locations of the passband and stopband edges. Furthermore, the proposed classes of FRM filters can, as special cases, realize efficient Mth-band FRM linear-phase FIR interpolation and decimation filters for all values of M. Previously, only half-band (M = 2) FRM linear-phase FIR filters have appeared in the literature. The paper includes design techniques suitable for the new filters and design examples illustrating their efficiency.     

Two-Channel FIR Filter Banks Utilizing the FRM Approach

The frequency-response masking (FRM) approach has been introduced as a means of generating narrow transition band linear-phase finite impulse response (FIR) filters with a low arithmetic complexity. This paper proposes an approach for synthesizing two-channel maximally decimated FIR filter banks utilizing the FRM technique. For this purpose, a new class of FRM filters is introduced. Filters belonging to this class are used for synthesizing nonlinear-phase analysis and synthesis filters for two types of two-channel filter banks. For the first type, there exist no phase distortion and aliasing errors, but this type suffers from a small amplitude distortion as for the well-known quadrature mirror filter (QMF) banks. Compared to conventional QMF filter banks, the proposed banks lower significantly the overall arithmetic complexity at the expense of a somewhat increased overall filter bank delay in applications demanding narrow transition bands. For the second type, there are also small aliasing errors, allowing one to reduce the arithmetic complexity even further. Efficient structures are introduced for implementing the proposed filter banks, and algorithms are described for maximizing the stopband attenuations of the analysis and synthesis filters in the minimax sense subject to the given allowable amplitude and/or aliasing errors. Examples are included illustrating the benefits provided by the proposed filter banks.     

Complexity Reduction By Decoupling The Masking Filters From The Bandedge Shaping Filter In The FRM Technique

In the frequency-response masking (FRM) approach, the complexity of two masking filters is heavily dependent on the interpolation factor and the cutoff frequencies of the bandedge shaping filter. In this paper, we propose a novel structure that decouples the masking filters from the bandedge shaping filter. The design equations together with the design procedures are presented. With the introduction of an additional decoupling stage, the complexity of the overall filter can be greatly reduced. Our example shows that more than 40% savings in the numbers of multipliers and adders can be achieved compared with the original FRM approach.     

Design of Sparse Filters for Channel Shortening

Channel shortening equalizers are used in acoustics to reduce reverberation, in error control decoding to reduce complexity, and in communication receivers to reduce inter-symbol interference. The cascade of a channel and channel shortening equalizer ideally produces an overall impulse response that has most of its energy compacted into fewer adjacent samples. Once designed, channel shortening equalizers filter the received signal on a per-sample basis and need to be adapted or re-designed if the channel impulse response changes significantly. In this paper, we evaluate sparse filters as channel shortening equalizers. Unlike conventional dense filters, sparse filters have a small number of non-contiguous non-zero coefficients. Our contributions include (1) proposing optimal and sub-optimal low complexity algorithms for sparse shortening filter design, and (2) evaluating impulse response energy compaction vs. design and implementation stage computational complexity tradeoffs for the proposed algorithms. We apply the proposed equalizer design procedures to (1) asymmetric digital subscriber line channels and (2) underwater acoustic communication channels. Our simulation results utilize measured channel impulse responses and show that sparse filters are able to achieve the same channel energy compaction with half as many coefficients as dense filters.     

Reconfigurable channel filtering and digital down conversion in optimal CSD space for software defined radio

Software defined radio (SDR) is a platform for using the same hardware to support multiple wireless communication standards. The channelizer in the SDR is used to separate the channels from the wideband signal. The features required for using the same hardware to switch between different standards are reconfigurability and low complexity. To achieve this, we use a variable bandwidth filter (VBF) with facility for the reduction and enhancement of the bandwidth, without changing the filter coefficients. This paper proposes an optimal and multiplier-less implementation of a baseband channel filter for supporting the bandwidth requirements of various wireless communication standards. This paper also discusses the concept for using it as a multi-band SDR channelizer for the direct conversion of signals from a wideband input to the baseband signal without using an intermediate stage. Frequency response masking (FRM) filter with continuous coefficients is designed and modified harmony search algorithm is used for finding the optimal canonic signed digit representation for the multiplier-less implementation. This reduces the complexity and power consumption. The VBF with the optimized FRM filter is evaluated for the selectable bandwidths starting from 1.25 to 8 MHz, which covers the bandwidth requirements of a wide range of wireless communication standards.     

基于神经网络的FRM滤波器优化设计研究

以频率响应屏蔽(FRM)技术为基础,提出了一种基于神经网络的窄过渡带FIR数字滤波器的优化设计新方法.该算法主要通过使频率响应平方误差函数最小化来获得FRM滤波器系数.文中详细介绍了基于神经网络的基本FRM滤波器和多层FRM滤波器的设计算法及设计步骤,证明了该算法的稳定性定理,给出了仿真实例,并与已有的设计方法进行了比较,设计结果表明用该方法设计的窄过渡带FIR数字滤波器性能更为优越.     

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.     

Synthesis of Wideband Linear-Phase FIR Filters with a Piecewise-Polynomial-Sinusoidal Impulse Response

A method is presented to synthesize wideband linear-phase finite-impulse-response (FIR) filters with a piecewise-polynomial-sinusoidal impulse response. The method is based on merging the earlier synthesis scheme proposed by the authors to design piecewise-polynomial filters with the method proposed by Chu and Burrus. The method uses an arbitrary number of separately generated center coefficients instead of only one or none as used in the method by Chu–Burrus. The desired impulse response is created by using a parallel connection of several filter branches and by adding an arbitrary number of center coefficients to form it. This method is especially effective for designing Hilbert transformers by using Type 4 linear-phase FIR filters, where only real-valued multipliers are needed in the implementation. The arithmetic complexity is proportional to the number of branches, the common polynomial order for each branch, and the number of separate center coefficients. For other linear-phase FIR filter types the arithmetic complexity depends additionally on the number of complex multipliers. Examples are given to illustrate the benefits of this method compared to the frequency-response masking (FRM) technique with regard to reducing the number of coefficients as well as arithmetic complexity.     

Design of high-resolution cosine-modulated transmultiplexers with sharp transition band

Due to the growing importance of multichannel modulation, there has been great interest in the design of high-performance transmultiplex systems. In this paper, a new cosine-modulated transmultiplex structure is proposed based on a prototype filter designed with the frequency-response masking (FRM) approach. This new structure leads to substantial reduction in the computational complexity (number of multiplications per output sample) of the prototype filters having sharp transition band and equivalently small roll-off values. The relation between the interpolation factor used in the FRM prototype filter and the decimation factor in the subbands leads to distinct structures. Examples included indicate that the reduction in computational complexity can be higher than 50% of the current state-of-art designs, whereas the reduction on the number of distinct coefficients of the prototype filter can be reduced even further (over 75%). As a result, the proposed approach allows the design of very selective subfilters for transmultiplexes with a very large number of subchannels.     

Design of Computationally Efficient Sharp FIR Filter Utilizing Modified Multistage FRM Technique for Wireless Communications Systems

Modern wireless communications gadgets demand multi-standard communications facilities with least overlap between different input radio channels. A sharp digital filter of extremely narrow transition-width with lower stop band ripples offers alias-free switching among the preferred frequency bands. A computationally competent low pass filter (LPF) structure based on the multistage frequency response masking (FRM) approach is proposed for the design of sharp finite impulse response (FIR) filters which are suitable for wireless communications applications. In comparison of basic FRM with other existing multistage FRM structures, the proposed structure has a narrow transition bandwidth and higher stop band attenuation with significant reduction in terms of the number of computational steps. A design example is incorporated to demonstrate the efficiency of the proposed approach. Simulation results establish the improvement of the proposed scheme over other recently published design strategies.     

On Low-Delay Frequency-Response Masking FIR Filters

This paper offers two main contributions to the theory of low-delay frequency-response masking (FRM) finite impulse response (FIR) filters. First, a thorough investigation of the low-delay FRM FIR filters and their subfilters or three different structures, referred to as narrow-, wide-, and middle-band filter structures, is given. The investigation includes discussions on delay distribution over the subfilters as well as estimation of the optimal periodicity of the periodic model filter. Second, systematic design procedures are given, with explicit formulas for distribution of the ripples and the delay to the subfilters. For each of the three structures, two design procedures are given that include joint optimization of the subfilters. The first proposal uses partly linear-phase FIR subfilters and partly low-delay FIR subfilters. Thus, it has a lower arithmetic complexity compared to the second proposal, which has exclusively low-delay FIR subfilters. The second proposal is instead more flexible and can handle a broader range of specifications. The design procedures result in low-delay FIR filters with a lower arithmetic complexity compared to previous results, for specifications with low delay and narrow transition band.     

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 unified approach to multistage frequency-response masking filter design using the WLS technique

This paper presents a unified approach to the optimal design of sharp linear-phase finite-impulse-response (FIR) digital filters synthesized using the multistage frequency-response masking (FRM) technique. In this approach, the design of a k-stage FRM filter is achieved in a recursive manner. The minimax design problem arising at each step of the synthesis process is converted into a corresponding weighted least-squares (WLS) problem. The WLS problem is highly nonlinear with respect to the coefficients of the filter. Consequently, it is decomposed into several linear least-squares (LS) problems, each of which can be solved analytically. It is then solved iteratively by using an alternating variable approach. Numerical design examples are included to demonstrate the effectiveness of the method.     

Optimal Design of Frequency-Response Masking Filters With Reduced Group Delays

In this paper, a new method for the design of optimal finite-impulse response frequency-response masking (FRM) filters with reduced passband group delays is proposed. To meet the prescribed magnitude response and group delay, the proposed design method takes into account both the magnitude error and the group delay error. The key step is the derivation of the group delay and its gradient with respect to the filter coefficients, based on which an explicit group delay constraint is formulated. By incorporating the group delay constraint into the overall optimization, FRM filters with better approximation to the prescribed reduced group delay can be obtained in comparison with a recent method by Lu and Hinamoto in 2003, as illustrated by two design examples.      

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.     

基于可重构FPGA技术的自适应FIR滤波器的实现

有限冲激响应(FIR)滤波器设计遇到的难题是滤波要进行大量乘法运算,即使是在全定制的专用集成电路中也会导致过大的面积与功耗.对于用硬件实现系数是常量的专用滤波器,可以通过分解系数变为应用加、减和移位而实现乘法.FIR滤波器的复杂性主要由用于系数乘法的加法器/减法器的数量决定.而对于自适应FIR滤波器,大多数场合下可用数字信号处理器(DSP)或CPU通过软件编程的方法来实现,但是对于要求高速运算的场合,VLSI实现是很好的选择.基于这一考虑,可以用符号数的正则表示(CSD)码表示系数, 再利用可重构现场可编程门阵列(FPGA)技术实现.可重构结构的应用,能保证系统的其余部分同时处于运行状态时实现FIR滤波器系数的更新.文中利用CSD码和可重构思想,提出了用FPGA实现自适应FIR滤波器的一种方案.