Throughput Maximizing FIR Filterbank For MIMO LTI Wireline Channels
FIR高通滤波器设计报告

新疆大学科学与技术学院MATLAB课程设计报告题目:FIR高通滤波器专业班级通信08-1学生姓名学生学号提交日期 2011年12月 30日目录一、设计目的(FIR高通滤波器设计) (1)二、设计要求 (1)三、设计内容 (1)3.1 FIR 滤波器设计原理 (1)3.1.1数字滤波器的设计原理 (1)3.1.2 FIR滤波器的基本结构 (1)3.1.3 FIR滤波器的主要特点 (2)3.1.4 FIR滤波器设计方法及MATLAB实现 (2)3.1.5窗函数选择与高通滤波器指标转换 (2)3.1.6滤波器的MATLAB实现 (3)3.2 程序代码 (3)3.3仿真结果与分析 (4)四、本设计改进与建议 (5)五、总结 (5)六、主要参考文献 (5)一、设计目的(FIR 高通滤波器设计) 二、设计要求在MATLAB 课程中,熟悉软件的使用以及FIR 滤波器的设计原理,锻炼程序的设计能力和整体思想三、设计内容1)产生一个多频信号,设计FIR 高通滤波器消除其中一些成分。
2)设计FIR 滤波器:阻带边缘频率10KHz,通带边缘频率22KHz ,阻带衰减75dB ,采样频率50KHz 。
3.1 FIR 滤波器设计原理FIR 高通滤波器设计3.1.1数字滤波器的设计原理数字滤波器的设计问题就是寻找一组系数ai 和bi ,使得其性能在某种意义上逼近所要求的特性。
如果在s 平面上去逼近,就得到模拟滤波器,如果在z 平面上去逼近,就得到数字滤波器。
数字滤波器是将输入的信号序列,按规定的算法进行处理从而得到所期望的输出序列。
一个线性位移不变系统的输出序列y(n)和输入序列x(n)之间的关系,应该满足常系数线性差分方程x (n )为输入序列,y (n )为输出序列,ai 、bi 为滤波器系数,N 为滤波器的阶数。
3.1.2 FIR 滤波器的基本结构设h(n)(n =0,1,2⋯一1)为滤波器的冲激响应,输入信号为x(n),则FIR 滤波器就是要实现下列差分方程:对式(2.2.1)进行z 变换,可得FIR 滤波器的传递函数:FIR 滤波器的单位冲击响应h(n)是一个有限长序列。
Filter Bank Design

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Lecture-8 Filter Banks – Special Topics
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优化FIR数字滤波器的FPGA实现

收稿日期 : 2010 11 26 基金项目 : 福建省 科技厅 重点项 目 ( 2008I0019) ; 福建 省自 然基金 项目 ( 2009J01285) 和福州市科技项目( 2010 G 102)
2. 1
152 FIR 滤波器的差分方程可表示为 :
t- 1
现代电子技术
2011 年第 34 卷
Abstract: T he traditional F IR digital filter was modified w ith the concept of increasing the speed and reducing the area. Taking into account of the characteristics of FPGA, the Booth Radix 2 algorithm multiplier and the fast adder w ere desig ned in the combina tion with CSA adder and a tree structur e, and then the adder and multiplier were used successfully in the design o f FIR digital filter. T he coefficients of the filter were generated by Matlab. Simulation and synthesis results show that Booth alg orithm multiplier and ad der tree based on CSA alg orithm not only meet the performance r equirements of FIR digital filter but also optimize the area on the cir cuit and especially the speed, and the optimization is mo re evident w hen there are a large amount of data. Keywords: M atlab; Bo oth alg or ithm; CSA alg or ithm; ISE
外文用改进的窗函数设计FIR数字滤波器

第31卷第6期2010年12月衡阳师范学院学报Journal o f Heng yang N o rmal U niv ersit yN o.6V o l.31Dec.2010用改进的窗函数设计FIR数字滤波器谭家杰,罗昌由,黄三伟,邓小辉(衡阳师范学院物理与电子信息科学系,湖南衡阳 421008)摘 要:窗函数设计FIR数字滤波器已多年,近年来,用优化技术设计数字滤波器十分流行,论文提出了一种新方法对窗函数进行改进,这种窗函数不同于以往的Hann窗、H amming窗、Blackman窗,它先将余弦序列线性组合为窗函数,然后根据FI R数字滤波器的特性对数字滤波器的幅度条件进行线性规划。
并且给出了改进窗函数的算法。
最后,用该方法设计出了F IR数字滤波器的仿真实例,并与用Hamming窗、Blackman窗方法设计的FIR滤波器进行对比,仿真结果表明用该方法设计的滤波器满足设计规格。
关键词:改进的窗函数;FIR数字滤波器;窗函数;线性规划中图分类号:T N713+ 7文献标志码:A文章编号:1673 0313(2010)06 0031 040 引 言FIR数字滤波器的设计方法主要有:窗函数法、频率采样法和切比雪夫等波纹逼近法[1 4]。
窗函数法是设计FIR数字滤波器最常用、最简单的方法[4 5]。
窗函数法的实质是用截断理想冲激响应的方法来逼近所求的滤波器指标。
频率采样法是一种优化设计方法,其缺点是设计时使用的变量仅限于过渡带上的几个采样值,截止频率不容易控制[3]。
切比雪夫等波纹逼近法是一种优化设计,但是存在计算复杂,计算量较大的缺点[1 2]。
窗函数法设计简单,有闭合形式的公式,因而很实用。
缺点是通带、阻带的截止频率不容易控制[2 3]。
数字滤波器的好坏取决于窗函数的选取,窗函数法设计的关键是:选择合适的窗函数,选择合适的阶数,改善数字滤波器的幅频特性,减少Gibbs 现象,解决收敛问题[1 2]。
FIR滤波器设计

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FIR数字滤波器的设计等波纹最佳逼近法

FIR 数字滤波器的设计--等波纹最佳逼近法一、等波最佳逼近的原理简介等波纹最佳逼近法是一种优化设计法,即最大误差最小化准则,它克服了窗函数设计法和频率采样法的缺点,使最大误差(即波纹的峰值)最小化,并在整个逼近频段上均匀分布。
用等波纹最佳逼近法设计的FIR 数字滤波器的幅频响应在通带和阻带都是等波纹的,而且可以分别控制通带和阻带波纹幅度,这就是等波纹的含义。
最佳逼近是指在滤波器长度给定的条件下,使加权误差波纹幅度最小化。
与窗函数设计法和频率采样法比较,由于这种设计法使滤波器的最大逼近误差均匀分布,所以设计的滤波器性能价格比最高。
阶数相同时,这种设计法使滤波器的最大逼近误差最小,即通带最大衰减最小,阻带最小衰减最大;指标相同时,这种设计法使滤波器阶数最低。
等波纹最佳逼近法的设计思想 。
用)(ωd H 表示希望逼近的幅度特性函数,要求设计线性相位FIR 数字滤波器时,)(ωd H 必须满足线性相位约束条件。
用()ωH 表示实际设计的滤波器的幅度特性函数。
定义加权误差函数()ωε为()()()()[]ωωωωεH H W d -=式中,()ωW 为幅度误差加权函数,用来控制不同频带(一般指通带和阻带)的幅度逼近精度。
等波纹最佳逼近法的设计在于找到滤波器的系数向量()n h ,使得在通带和阻带内的最大绝对值幅度误差()ωε为最小,这也就是最大误差最小化问题。
二、等波纹逼近法设计滤波器的步骤和函数介绍1.根据滤波器的设计指标的要求:边界频率,通带最大衰减,阻带最大衰等估计滤波器阶数n ,确定幅度误差加权函数()ωW2.采用Parks-McClellan 算法,获得所设计滤波器的单位脉冲响应()n h实现FIR 数字滤波器的等波纹最佳逼近法的MATLAB 信号处理工具函数为firpm 和firpmord 。
firpm 函数采用数值分析中的多重交换迭代算法求解等波纹最佳逼近问题,求的满足等波纹最佳逼近准则的FIR 数字滤波器的单位脉冲响应()n h 。
Filter bank
专利名称:Filter bank发明人:GOECKLER, HEINZ, DIPL.-ING.申请号:EP88105684.0申请日:19880409公开号:EP0290790B1公开日:19970730专利内容由知识产权出版社提供摘要:The invention relates to a filter bank for frequency demultiplexing or multiplexing L channel signals, for L > 0 and L being a natural number, consisting of L transversal filters (non-recursive filters, filters with a finite duration of the pulse response (FIR)), with complex-valued coefficients, for real-valued input signals and with complex-valued output signals or conversely, with a sampling rate which is reduced or increased by the factor M at the output, and is characterised by the fact that at least one of theL filters is implemented by cascading at least two filters having in each case complex-valued coefficients, that in each of these filters the sampling rate is reduced or increased, respectively, by the factor Mq > 1, with… …… that the input filters of the cascade in each case receive a real-valued input signal and output a complex output signal and the other filters of the cascade in each case receive a complex-valued input signal and output a complex-valued output signal (Figure 3) or, respectively, that the first Q-1 filters of the cascade in each case receive a complex-valued input signal and output a complex-valued output signal and the output filters of the cascade in each case receive a complex-valued input signal and output a real-valued output signal.申请人:ROBERT BOSCH GMBH,BOSCH GMBH ROBERT,ROBERT BOSCH GMBH地址:DE国籍:DE更多信息请下载全文后查看。
FIR滤波器文献综述
FIR滤波器⽂献综述⽂献综述⼀.FIR滤波器的概念,原理滤波器(filter),是⼀种⽤来消除⼲扰杂讯的器件,将输⼊或输出经过过滤⽽得到纯净的直流电。
对特定频率的频点或该频点以外的频率进⾏有效滤除的电路,就是滤波器,其功能就是得到⼀个特定频率或消除⼀个特定频率。
滤波器分为有源滤波和⽆源滤波。
主要作⽤是:让有⽤信号尽可能⽆衰减的通过,对⽆⽤信号尽可能⼤的衰减。
数字滤波器具有稳定、重复性好、适应性强、性能优异、线性相位等优点。
数字滤波器以冲激响应延续长度可分为两类:FIR 滤波器(有限冲激响应滤波器)、IIR滤波器(⽆限冲激响应滤波器)。
FIR、IIR是常⽤的数字滤波器。
特点是随着阶数的增加,滤波器过渡带越来越窄,也即矩形系数越来越⼩。
FIR是线性相位的,⽆论多少阶,在通带内的信号群时延相等,也即⽆⾊散,对于PSK这类信号传输尤为重要,IIR通常是⾮线性的,但是⽬前也有准线性相位设计⽅法得到IIR数字滤波器的系数,其结果是使得通带内的相位波动维持在⼀个⼯程可接受的范围内。
IIR ⽐FIR最⼤的优点是达到同样的矩形系数所需的阶数少,往往5阶的IIR滤波器就可以⽐拟数⼗上百阶的FIR滤波器。
但是另⼀⽅⾯,FIR滤波器的系数设计⽅法很多,最普遍的是加窗,种类繁多的窗函数可以得到各种你所需要的通带特性。
FIR(Finite Impulse Response)滤波器:有限长单位冲激响应滤波器,是数字信号处理系统中最基本的元件,它可以在保证任意幅频特性的同时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,因⽽滤波器是稳定的系统。
FIR 滤波器在通信、图像处理,模式识别等领域都有着⼴泛的应⽤。
⽬前,FIR滤波器的硬件实现有以下⼏种⽅式:1、数字集成电路FIR滤波器⼀种是使⽤单⽚通⽤数字滤波器集成电路,这种电路使⽤简单,但是由于字长和阶数的规格较少,不易完全满⾜实际需要。
虽然可采⽤多⽚扩展来满⾜要求,但会增加体积和功耗,因⽽在实际应⽤中受到限制。
关于MATLAB实现FIR滤波器优化设计方法-英文论文
FIR Filter Design T echniquesAbstractThis report deals with some of the techniques used to design FIR filters. In the beginning, the windowing method and the frequency sampling methods are discussed in detail with their merits and demerits. Different optimization techniques involved in FIR filter design are also covered, including Rabiner‟s method for FIR filter design. These optimization techniques reduce the error caused by frequency sampling technique at the non-sampled frequency points. A brief discussion of some techniques used by filter design packages like Matlab are also included.IntroductionFIR filters are filters having a transfer function of a polynomial in z and is an all-zero filter in the sense that the zeroes in the z-plane determine the frequency response magnitude characteristic.The z transform of a N-point FIR filter is given by(1)FIR filters are particularly useful for applications where exact linear phase response is required. The FIR filter is generally implemented in a non-recursive way which guarantees a stable filter.FIR filter design essentially consists of two parts(i) approximation problem(ii) realization problemThe approximation stage takes the specification and gives a transfer function through four steps. They are as follows:(i) A desired or ideal response is chosen, usually in the frequency domain.(ii) An allowed class of filters is chosen (e.g.the length N for a FIR filters).(iii) A measure of the quality of approximation is chosen.(iv) A method or algorithm is selected to find the best filter transfer function.The realization part deals with choosing the structure to implement the transfer function which may be in the form of circuit diagram or in the form of a program.There are essentially three well-known methods for FIR filter design namely:(1) The window method(2) The frequency sampling technique(3) Optimal filter design methodsThe Window MethodIn this method, [Park87], [Rab75], [Proakis00] from the desired frequency response specification Hd(w), corresponding unit sample response hd(n) is determined using the following relation(2)(3)In general, unit sample response hd(n) obtained from the above relation is infinite in duration, so it must be truncated at some point say n= M-1 to yield an FIR filter of length M (i.e. 0 to M-1). This truncation of hd(n) to length M-1 is same as multiplying hd(n) by the rectangular window defined asw(n) = 1 0≦n≦M-1 (4)0 otherwiseThus the unit sample response of the FIR filter becomesh(n) = hd(n) w(n) (5)= hd(n) 0≦n≦M-1= 0 otherwiseNow, the multiplication of the window function w(n) with hd(n) is equivalent to convolution of Hd(w) with W(w), where W(w) is the frequency domain representation of the window function(6)Thus the convolution of Hd(w) with W(w) yields the frequency response of the truncated FIR filter(7)The frequency response can also be obtained using the following relation(8)But direct truncation of hd(n) to M terms to obtain h(n) leads to the Gibbs phenomenon effect which manifests itself as a fixed percentage overshoot and ripple before and after an approximated discontinuity in the frequency response due to the non-uniform convergence of the fourier series at a discontinuity.Thus the frequency response obtained by using (8) contains ripples in the frequency domain. In order to reduce the ripples, instead of multiplying hd(n) with a rectangular window w(n), hd(n) is multiplied with a window function that contains a taper and decays toward zero gradually, instead of abruptly as it occurs in a rectangular window. As multiplicationof sequences hd(n) and w(n) in time domain is equivalent to convolution of Hd(w) and W(w) in the frequency domain, it has the effect of smoothing Hd(w).The several effects of windowing the Fourier coefficients of the filter on the result of the frequency response of the filter are as follows:(i) A major effect is that discontinuities in H(w) become transition bands between values on either side of the discontinuity.(ii) The width of the transition bands depends on the width of the main lobe of the frequency response of the window function, w(n) i.e. W(w).(iii) Since the filter frequency response is obtained via a convolution relation , it is clear that the resulting filters are never optimal in any sense.(iv) As M (the length of the window function) increases, the mainlobe width of W(w) is reduced which reduces the width of the transition band, but this also introduces more ripple in the frequency response.(v) The window function eliminates the ringing effects at the bandedge and does result in lower sidelobes at the expense of an increase in the width of the transition band of the filter.Some of the windows [Park87] commonly used are as follows:1. Bartlett triangular window:W(n)=2(n+1)/N+1 n=0,1,2…….,(N-1)/2 (9) =2-2(n+1)/N+1 n=(N-1)/2,……,N-1= 0 otherwise2 Generalized cosine windows(Rectangular, Hanning, Hamming and Blackman)W(n)=a-bcos(2p(n+1)/(N+1))+ccos(4p(n+1)/(N+1)) n=0,1….N-1 (10) = 0 otherwise3.Kaiser window with parameter β:(11)The general cosine window has four special forms that are commonly used. These are determined by the parameters a,b,cTABLE IV alue of coefficients for a,b and c from [Park87]Window a b cRetangular 1 0 0Hanning 0.5 0.5 0Hamming 0.54 0.46 0Blackman 0.42 0.5 0.08The Bartlett window reduces the overshoot in the designed filter but spreads the transition region considerably.The Hanning,Hamming and Blackman windows use progressively more complicated cosine functions to provide a smooth truncation of the ideal impulse response and a frequency response that looks better. The best window results probably come from using the Kaiser window, which has a parameter . that allows adjustment of the compromise between the overshoot reduction and transition region width spreading.The major advantages of using window method is their relative simplicity as compared to other methods and ease of use. The fact that well defined equations are often available for calculating the window coefficients has made this method successful.There are following problems in filter design using window method:(i) This method is applicable only if Hd(w) is absolutely integrable i.e only if (2) can be evaluated. When Hd(w) is complicated or cannot easily be put into a closed form mathematical expression, evaluation of hd(n) becomes difficult.(ii) The use of windows offers very little design flexibility e.g. in low pass filter design, the passband edge frequency generally cannot be specified exactly since the window smears the discontinuity in frequency. Thus the ideal LPF with cut-off frequency fc, is smeared by the window to give a frequency response with passband response with passband cutoff frequency f1 and stopband cut-off frequency f2.(iii) Window method is basically useful for design of prototype filters like lowpass,highpass,bandpass etc. This makes its use in speech and image processing applications very limited.The Frequency Sampling TechniqueIn this method, [Park87], [Rab75], [Proakis00] the desired frequency response is provided as in the previous method. Now the given frequency response is sampled at a set of equally spaced frequencies to obtain N samples. Thus , sampling the continuous frequency response Hd(w) at N points essentially gives us the N-point DFT of Hd(2pnk/N). Thus by using the IDFT formula, the filter co-efficients can becalculated using the following formula(12)Now using the above N-point filter response, the continuous frequency response is calculated as an interpolation of the sampled frequency response. The approximation error would then be exactly zero at the sampling frequencies and would be finite in frequencies between them. The smoother the frequency response being approximated, the smaller will be the error of interpolation between the sample points.One way to reduce the error is to increase the number of frequency samples [Rab75]. The other way to improve the quality of approximation is to make a number of frequency samples specified as unconstrained variables. The values of these unconstrained variables are generally optimized by computer to minimize some simple function of the approximation error e.g. one might choose as unconstrained variables the frequency samples that lie in a transition band between two frequency bands in which the frequency response is specified e.g. in the band between the passband and the stopband of a low pass filter.There are two different set of frequencies that can be used for taking the samples. One set of frequency samples are at fk = k/N where k = 0,1,….N-1. The other set of uniformly spaced frequency samples can be taken at fk =(k+1/2)/N for k = 0,1,….N-1.The second set gives us the additional flexibility to specify the desired frequency response at a second possible set of frequencies. Thus a given band edge frequency may be closer to type-II frequency sampling point that to type-I in which case a type-II design would be used in optimization procedure.In a paper by Rabiner and Gold [Rabi70], Rabiner has mentioned a technique based on the idea of frequency sampling to design FIR filters. The steps involved in this method suggested by Rabiner are as follows:(i) The desired magnitude response is provided along with the number of samples,N . Given N, the designer determines how fine an interpolation will be used.(ii) It was found by Rabiner that for designs they investigated, where N varied from 15 to 256, 16N samples of H(w) lead to reliable computations, so 16 to 1 interpolation was used.(iii) Given N values of Hk , the unit sample response of filter to be designed, h(n) is calculated using the inverse FFT algorithm.(iv) In order to obtain values of the interpolated frequency response two procedureswere suggested by Rabiner.They are(a) h(n) is rotated by N/2 samples(N even) or (N-1)/2 samples for N odd to remove the sharp edges of impulse response, and then 15N zero-valued samples are symmetrically placed around the impulse response.(b) h(n) is split around the N/2nd sample, and 15N zero-valued samples are placed between the two pieces of the impulse response.(v) The zero augmented sequences are transformed using the FFT algorithm to give the interpolated frequency responses.Merits of frequency sampling technique(i) Unlike the window method, this technique can be used for any given magnitude response.(ii) This method is useful for the design of non-prototype filters where the desired magnitude response can take any irregular shape.There are some disadvantages with this method i.e the frequency response obtained by interpolation is equal to the desired frequency response only at the sampled points. At the other points, there will be a finite error present.Optimal Filter Design MethodsMany methods are present under this category. The basic idea in each method is to design the filter coefficients again and again until a particular error is minimized. The various methods are as follows:(i)Least squared error frequency domain design(ii)Nonlinear equation solution for maximal ripple FIR filters(iii)Polynomial interpolation solution for maximal ripple FIR filtersLeast squared error frequency domain designAs seen in the previous method of frequency sampling technique there is no constraint on the response between the sample points, and poor results may be obtained.The frequency sampling technique is more of an interpolation method rather than an approximation method. This method [Rab75], [Parks87] controls the response between the sample points by considering a number of sample points larger than theorder of the filter.The purpose of most filters is to separate desired signals from undesired signals or noise. As the energy of the signal is related to the square of the signal, a squared error approximation criterion is appropriate to optimize the design of the FIR filters.The frequency response of the FIR filter is given by (8) for a N-point FIR filter. An error function is defined as follows(13) where wk=(2*pi*k)/L and Hd(wk) are L samples of the desired response, which is the error measure as a sum of the squared differences between the actual and desired frequency response over a set of L frequency samples. The method consists of the following steps:(i) First …L‟ samples from the continuous frequency response are taken, where L>N(length of the impulse response of filter to be designed)(ii) Then using the following formula(14) the L-point filter impulse response is calculated.(iii) Then the obtained filter impulse response is symmetrically truncated to desired length N.(iv) Then the frequency response is calculated using the following relation(15) (v) The magnitude of the frequency response at these frequency points for wk =(2k)/L will not be equal to the desired ones , but the overall least square error will be reduced effectively this will reduce the ripple in the filter response.To further reduce the ripple and overshoot near the band edges, a transition region will be defined with a linear transfer function. Then the L frequency samples are taken at wk =(2*pi*k)/L using which the first N samples of the filter are calculated using the above ing this method, reduces the ripple in the interpolated frequency response.Nonlinear Equation solution for maximal ripple FIR filters The real part of the frequency response of the designed FIR filter can be written as a(n)cos(wn) [Rab75] where limits of summation and a(n) vary according to the type of the filter. The number of frequencies at which H(w) could attain an extremum is strictly a function of the type of the linear phase filter i.e. whether length N of filter isodd or even or filter is symmetric or anti-symmetric. At each extremum, the value of H(w) is predetermined by a combination of the weighting function W(w), the desired frequency response, and a quantity that represents the peak error of approximation distributing the frequencies at which H(w) attains an extremal value among the different frequency bands over which a desired response was being approximated. Since these filters have the maximum number of ripples, they are called maximal ripple filters.This method is as follows:At each of the Ne unknown external frequencies, E(w) attains the maximum value of either and E(w) or equivalently H(w) has zero derivative. Thus two Ne equations of the formH(wi)= ±δ/W(w) + D(wi) (16)d/dw {H(wi)} at w=wi=0 (17) are obtained.These equations represent a set of 2Ne nonlinear equations in two Ne unknowns, Ne impulse response coefficients and Ne frequencies at which H(w) obtains the extremal value. The set of two Ne equations may be solved iteratively using nonlinear optimiation procedure.An important thing to note is that here the peak error (δ) is a fixed quantity and is not minimized by the optimization scheme. Thus the shape of H(w) is postulated apriori and only the frequencies at which H(w) attains the extremal values are unknown.The disadvantage of this method is that the design procedure has no way of specifying band edges for the different frequency bands of the filter. Thus the optimization algorithm is free to select exactly where the bands will lie.Polynomial Interpolation Solution for Maximal Ripple FIR filters This algorithm [Rab75] is basically an iterative technique for producing a polynomial H(w) that has extrema of desired values. The algorithm begins by making an initial estimate of the frequencies at which the extrema in H(w) will occur and then uses the well-known Lagrange interpolation formula to obtain a polynomial that alternatively goes through the maximum allowable ripple values at these frequencies. It has been experimentally found that the initial guess of extremal frequencies does not affect the ultimate convergence of the algorithm but instead affects the number ofiterations required to achieve the desired result.Let us consider the case of design of a low pass filter using the above method.Fig. 1. Iterative solution for a maximum ripple lowpass filter from [Rab75]The Fig. 1 shows the response of a lowpass filter with N = 11.The number of extremal frequencies i.e. the frequencies where ripples occur are 6 in this case. They are divided into 3 passband extrema and 3 stopband extrema. The filled dots indicate the initial guess as to the extremal frequencies of H(w). The solid line is the initial Lagrange polynomial obtained by choosing polynomial coefficients so that the values of the polynomial at the guessed set of frequencies are identical to the assigned extreme values. But this polynomial has extrema that exceeds the specified maxima values. The next stage of the algorithm is to locate the frequencies at which the extrema of the first Lagrange interpolation occur. These frequencies are now used as the new frequencies for which the extrema of the filter response occur. This second set of frequencies are indicated by open dots in Fig. 1. Now similarly the new set of frequencies are taken as those frequencies where the maximum exceeds the specified maxima. Thus the method is completely iterative in nature.ConclusionsThe report has described the various techniques involved in the design of FIR filters. Every method has its own advantages and disadvantages and is selected depending on the type of filter to be designed.The window method is basically used for the design of prototype filters like the low-pass, high-pass, band-pass etc. They are not very suitable for designing of filters with any given frequency response. On the other hand, the frequency sampling technique is suitable for designing of filters with a given magnitude response. The ideal frequency response of the filter is approximated by placing appropriate frequency samples in the z -plane and then calculating the filter co-efficients using the IFFT algorithm.The disadvantage of the frequency sampling technique was that the frequency response gave errors at the points where it was not sampled. In order to reduce these erros the different optimization technique for FIR filter design were presented wherein the remaining frequency samples are chosen to satisfy an optimization criterion. An appendix consisting of the filter design methods used by the software package Matlab is also presented.References[Park87] T.W. Parks and C.S. Burrus, Digital Filter Design. New Y ork:Wiley,1987 [Rab75] L.R. Rabiner and B. Gold, Theory and Applications of Digital Signal Processing. New Jersey: Prentice-Hall, 1975[Proakis00] J.G.Proakis and D.G. Manolakis, Digital Signal Processing Principles, Algorithms and Applications New Delhi: Prentice-Hall, 2000[Rabi70] L.R. Rabiner, B. Gold and C.A. McGonegal, “An approach to the Approximation Problem for Nonrecursive Digital Filters,” IEEE Trans. Audio and Electroacoustics, vol. AU-18, pp. 83-105,June 1970.[Mat02]/access/helpdesk/help/toolbox/signal/signal.shtml Signal Processing Toolbox,The Mathworks,Inc., accessed October 25,2002。
FIR(有限冲激响应)滤波器9页
实验 3: 设计 FIR(有限冲激响应)滤波器针对 Spartan-3E 开发套件介绍在这个实验里, 将向你展示通过系统发生器的FIR 和 FDATool模块来指定、模拟和实现FIR滤波器的方法. FDATool 模块被用来定义滤波器的阶数和系数, FIR模块被用作 Simulink 模拟以及在 FPGA中用 Xilinx ISE来实现设计.你也可以通过实际硬件来运行它以验证这个设计的功能.注意:在 c:\xup\dsp_flow\labs\labsolutions\lab3\ 目录下有完整的例子.目标在完成这个试验后, 你将能够:•用 FDATool 模块输入你的滤波器指数,在设计中使用产生的系统或把它储存在工作区•使用具有 FDATool 模块产生的系数的FIR模块, 并在Simulink中运行 bit-true 仿真•产生设计并用 Resource Estimator 模块和post-map 报告估计资源利用量设计描述你是Cyberdyne系统的一个dsp设计者. 你的公司正在调查用数字滤波器代替安防检测器中的模拟滤波器,以尝试提高性能和降低整个系统的成本. 这将使贵公司可以进一步渗入日益增长的安防市场. 一个单频取样滤波器设计如下:•Sampling Frequency (Fs)(采样频率) = 1.5 MHz•Fstop 1 = 270 kHz•Fpass 1 = 300 kHz•Fpass 2 = 450 khz•Fstop 2 = 480 kHz•Attenuation on both sides of the passband(双边通频带衰减) = 54 dB•Pass band ripple(通频带脉动) = 1因为灵活性和上市时间的原因,Cyberdyne已经选择FPGA来实现它.你的HDL设计经验是有限的. 因为你对MathWorks的产品比较熟悉,所以System Generator for DSP会是一个在FPGA中实现滤波器的优秀解决方案.你的经理 Miles Booth 已经要求你创建一个在即将完成的 Spartan-3E™ 原型板上实现的滤波器的原型. 这个原型必须尽可能快地完成,这是因为Aggressive Security 会议即将来临, 这个会议是业界今年最大的会议,我们不能错过它.你的经理已经提供了具有输入源和输出接收端的最初模型. 你的设计必须用随机输入源和来自DSP Blockset 的chirp 信号来仿真. 为分析滤波器的输出, 输入和输出信号在频谱示波器中显示. 频谱示波器用来比较在FPGA 中实现的定点 FIR 滤波器的录放频谱响应.两个不同的输入源用来仿真这个滤波器:• chirp 信号模块, 其扫描频率为 6 KHz 到 10 KHz ,不管其瞬时输出频率• 随机信号源发生器, 它输出范围在-1.9到1.9的均匀分布的随机信号. 因为均匀信号是有界的,它是驱动定点滤波器的较好的选择.设计流程这个试验由六个主要步骤组成. 在步骤1,你将用系统发生器的FDATool 模块来产生指定FIR 滤波器的系数. 在步骤2, 你将把这个系数与 FIR filter 模块联系起来. 步骤3要求你用两个已提供的输入源仿真这个设计并分析系数和输入信号宽度变化的影响. 在步骤4, 你将要添加一个转换模块以调整输出宽度,添加一个延时模块以提高效率,添加一个资源估计模块以估计资源占用量,从而完成整个设计. 在下一步将实现这个设计. 在每一步总的指示下面, 你会发现手把手的说明和图表,指出了实现总的指示的更多细节. 如果你对个别指导有信心, 可以略过手把手的说明,跳到下一步的总指示. 注意: 如果你当时不能完成这个实验, 你可以从Xilinx 大学计划网站 http://university.xilinx 下载实验文件产生FIR 滤波器所需的系数 步骤 1实验总流程: 从 Xilinx Blockset DSP blockset 中添加FDATool 模块到包含一个 DA FIR 滤波器的设计中. 按照下列要求用FDATool 模块产生FIR 滤波器的系数 • Sampling Frequency (Fs) (采样频率)= 1.5 MHz • Fstop 1 = 270 KHz• Fpass 1 = 300 KHz• Fpass 2 = 450 Khz• Fstop 2 = 480 KHz• Attenuation on both sides of the passband (双边通频带衰减) = 54 dB步骤 1: 产生 滤波器 系数步骤 2: 把系数与滤波器相关联步骤 3:仿真 滤波器 步骤 4: 完成整个 设计 Step 6: 估计资源 占用量 Step 5: 实现设计 Step 7: 执行 HW-in-the-•Pass band ripple(通频带脉动) = 1❶在 Matlab中, 改变目录为c:/xup/dsp_flow/labs/lab3/: 在命令行窗口键入cd c:/xup/dsp_flow/labs/lab3/ .❷从MATLAB控制台窗口打开bandpass_filter.mdl模块❸从Xilinx Blockset DSP添加FDATool模块到这个设计❹在FDATool Design Filter窗口 (图 3-1)输入下列滤波器参数•Response Type: Bandpass•Units: KHz•Sampling Frequency (Fs) = 1.5 MHz•Fstop 1 = 270 KHz•Fpass 1 = 300 KHz•Fpass 2 = 450 Khz•Fstop 2 = 480 KHz•Attenuation on both sides of the passband = 54 dB (Astop1 and Astop2 parameters)•Pass band ripple = 1 (Apass)图 3-1. 在 FDATool中设计一个滤波器.❺点击Design Filter按钮以确定滤波器指令频谱窗口将被更新并显示如图 3-2图 3-2. 所设计滤波器的幅频响应.1. 基于所定义的技术要求, 最小的滤波器阶数是多少?❻用File Save Session来储存fda格式文件coefficients.fda 中的系数注意: 这是可选的步骤. 这些系数对这个设计仍然可以利用. 如果你把参数储存在 fda-file中,你可以通过 FDATool 模块参数对话框来加载它们.❼用File Export导出工作区中的系数,其Numerator variable name为Num(图3-3)注意: 这将在你的 MATLAB 工作区添加Num变量. 对于一个 FIR 滤波器, Num 代表用在滤波器中的系数. 因为通过 FDATool 模块仍能利用这些系数,这也是一可选的步骤图 3-3. 在工作区中导出系数.❽在 MATLAB 控制台窗口键入Num来查看系数列表❾在 MATLAB 控制台窗口键入max(Num)以确定指出系数宽度和二进制小数点的最大系数值2. 填入与系数相关的下列信息最大值: ____________________最小值: 把系数与 FIR 滤波器相关联步骤 2 实验总流程: 从 Xilinx Blockset DSP library 添加FIR 滤波器模块并与产生的系数关联 ❶ 从 Xilinx Blockset DSP library 添加FIR (DA FIR v9_0)滤波器到设计中❷ 双击 Xilinx FIR Filter 模块 并在模块参数窗口 (图 3-4)中输入以下参数• Coefficients : xlfda_numerator(‘FDATool’)• Coefficient Structure : Inferred from Coefficients• Number of bits per Coefficients : 12• Binary Point for Coefficients : 11• Provide Valid Ports : unchecked图 3-4. FIR 滤波器模块参数.❸ 点击 OK 以接受设置❹ 如图 3-5所示连接设计的各个模块图 3-5. 设计用来仿真的FIR 滤波器模块.在Simulink 中仿真 FIR 滤波器 步骤 3实验总流程:设置样值输入为 FIX_8_6 ,输入采样周期为 1/1500000. 利用频谱示波器, 研究 chirp 和 noise 信号的输出. ❶ 双击 Gateway In 模块并设置格式为 FIX_8_6 ,采样周期为1/1500000❷ 选择 Chirp 信号为输入源并开始仿真如果你收到下列错误, 根据消息更新FIR 的等待时间, 并再次仿真. ❸ 调出示波器并验证一直衰减的FIR 滤波器的输出信号,这些信号应如图 3-7 和 图 3-8所示.图 3-7. 无衰减通频带 (频谱示波器).图 3-8. 禁带衰减 (频谱示波器).❹ 停止仿真❺ 选择 Random Source 并运行仿真图 3-10. 随机信号输入源 (频谱示波器).Step 1: Generate Filter Coeffici Step 2: Associat e Coeffici Step 3:Simulate the Filter Step 4: Complete the Step 6: Estimate Resource Step 5: Implemen t Design Step 7: Perform HW-in-Step 1: Generate Filter Coeffici Step 2: Associat e Coeffici Step 3:Simulate the Filter Step 4: Complete the Step 6: Estimate Resource Step 5: Implemen t Design Step 7: Perform HW-in-❻ 停止仿真完成 FIR 滤波器设计步骤 4实验总流程: 加入 convert 模块以得到 FIX_8_6 格式的输出,从而降低显示所需的动态量程. 加入 delay 部件以提高性能. 利用 Resource Estimator 估计设计所需的资源.❶ 从 Xilinx Blockset Basic Elements 库中添加一个Convert模块到 FIR 滤波器的输出 以使输出符合 FIX_8_6 格式并改变quantization 为Truncate , Overflow 为 Wrap❷ 从 Xilinx Blockset Basic Elements 库中添加一个延时部件到输出以实现流水线技术 并提高性能,流水线技术会在输出衰减器中得到实现❸ 确信 FIR size 被设置为 FIX_12_11 , Gateway In size 被设置为 FIX_8_6❺ 从 Xilinx Blockset Index 库中添加Resource Estimator 模块到设计中注意: 注意你的设计应如图 3-11所示.图 3-11. 完成的FIR 滤波器设计.实现 FIR 滤波器 步骤 5实验总流程:设置 FIR 硬件过采样率为 9, FIR 核等待时间为14, 然后运行仿真. 这将更新采样率为 7.407e-008. 按照以下技术要求用系统发生器产生代码. 在工程向导里打开 bandpass_filter.ise 工程, 综合并执行它.• Input : Width = FIX_8_6, Quantization = Truncate,Overflow = Wrap• Output Width: FIX_8_6• FIR Core Latency: 14• FIR Hardware Over-Sampling Rate: 9• FIR Coefficients: FIX_12_12• Compilation: HDL Netlist• Part: Spartan3e xc3s500e-4fg320• Synthesis Tool: XSTStep 1: Generate Filter Coeffici Step 2: Associat e Coeffici Step 3:Simulate the Filter Step 4: Complete the Step 6: Estimate Resource Step 5: Implemen t Design Step 7: Perform HW-in-Step 1: Generate Filter Coeffici Step 2: Associat e Coeffici Step 3:Simulate the Filter Step 4: Complete the Step 6: Estimate Resource Step 5: Implemen t Design Step 7: Perform HW-in-• Target Directory: c:/xup/dsp_flow/labs/lab3/ise• Create Testbench: Unchecked• Simulink System Period (sec): 7.407e-008•FPGA System Clock Period (ns): 20❶ 双击 FIR 模块并设置 Hardware Over-Sampling Rate 为 9 , latency 为 143. Hardware Over-Sampling Rate 设置为 9 后,滤波器的实现结果有何变化? 为何要把它设为 9 而不是 8?❷ 运行仿真. 如果采样率更新窗口出现, 接受采样率为 7.407e-008 并再次运行仿真❸ 双击系统发生器符号并设置以下参数• Compilation: HDL Netlist• Part: Spartan3e xc3s500e-4fg320• Synthesis Tool: XST• Target Directory: c:/xup/dsp_flow/labs/lab3/ise(or ./ise)• Create Testbench: Unchecked• Simulink System Period (sec): 7.407e-008• FPGA System Clock Period (ns): 20❹ 双击 Generate 按钮以产生设计❺ 在资源管理器中双击 bandpass_filter_cw.ise 以打开Project Navigator❹ 高亮突出顶层文件并双击执行4. 利用各种报告, 回答以下问题Slices 的数目:有无定时约束?实际时钟周期:5. 在选择连续执行的情况下, 滤波器的实际时钟是多少?用 Resource Estimator 估计资源利用率 步骤 6实验总流程: 用 Resource Estimator 模块和 post-map 报告, 估计此设计所用的资源. ❶ 双击 resource estimator 模块❷ 从下拉框中选择 Post-Map 并点击 Estimate 按钮.ISE 将自动通过综合和映射来运行并产生映射报告.? Step 1: Generate Filter Coeffici Step 2: Associat e Coeffici Step 3:Simulate the Filter Step 4: Complete the Step 6: Estimate Resource Step 5: Implemen t Design Step 7: Perform HW-in-6. 资源估计器的报告有什么内容?Slices 的数目:FFs 的数目:LUTs 数目:实现 Hardware-in-the-Loop 验证 步骤 7实验总流程: 利用系统发生器产生硬件电路并通过硬件电路板来验证设计. 通过Simulink 仿真这个设计. ❶ 储存模型为 bandpass_filter_hwcosim.mdl ❷• Compilation : Hardware Co-simulation xupsp3e_starter_kit• Synthesis Tool : XST• Target Directory : c:/xup/dsp_flow/labs/lab3/sp3e(or ./sp3e)• Create Testbench: Unchecked• Simulink System Period (sec): 7.407e-008❸ 点击 Generate 按钮,会出现一个对话框,显示编译处理的过程,如图 3-12图 3–12. 在命令窗口的编译过程.❹ 当发生器成功完成后, 一个新的 Simulink 库窗口将打开,并出现一个具有适当输入输出数的模块.图 3–13. 在新的 Simulink 窗口中打开的被编译模块.❹ 复制被编译模块到设计中并连结它如图 3-14所示图 3–14. 在循环仿真中完成硬件的设计准备工作.连接硬件板并通过Simulink 对设计仿真.❶ 把电源线与硬件板相连. 板子的LEDs 将闪亮.❷ 在板子与PC 间连接下载电缆❸ 在硬件协同仿真模块上双击并设置下载电缆为USB.❸在Simulink 窗口中点击 run 按钮 ( ) 以运行仿真. 配置文件将被下载,仿真将开始运行 ❹ 仿真结果将在输出示波器中显示,其中 Simulink 仿真器的输出显示在顶部,硬件输出显示在底部(Figure 3-15)Step 1: Generate Filter Coeffici Step 2: Associat e Coeffici Step 3:Simulate the Filter Step 4: Complete the Step 6: Estimate Resource Step 5: Implemen t Design Step 7: Perform HW-in-图 3–15. 仿真结果,Simulink 仿真器的输出显示在顶部.硬件输出显示在底部❺完成后关闭电源❻储存模块并关闭 MATLAB结论在这个试验里, 你学会了如何用 FDATool 创建滤波器系数并从FIR滤波器模块使用它们.学会使用Resource Estimator模块估计资源占用量.答案A1. 基于定义的技术要求, 最小的滤波器阶数是多少?922. 完成下列与系数相连的信息最大值: 0.1610最小值: -0.15413. Hardware Over-Sampling Rate 设置为 9 后,滤波器的实现结果有何变化? 为何要把它设为 9 而不是8?它指出滤波器可以连续实现. 每次输出会占用9个时钟周期. 设为9而不设为8是因为要实现系数的对称. 在对称的条件下, 可以在通过串行乘法器之前实现采样. 这实现了8位数相加,结果为9位数的执行结果 (位增长).4. 利用各种报告, 回答下列问题Slices的数目: 278有无时间约束? Yes实际的时钟周期: ~146 MHz5. 在你选择连续执行的情况下, 滤波器实际的时钟周期是多少?1.5MHz x 9 = 135 MHz6. 资源估计器的报告内容?Slices的数目: 282FFs的数目: 481LUTs的数目: 410希望以上资料对你有所帮助,附励志名言3条:1、宁可辛苦一阵子,不要苦一辈子。
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Throughput Maximizing FIR Filterbank For MIMOLTI Wireline ChannelsAhmed F.ShalashAnalog DevicesNew Jersey Design CenterSomerset,NJ08873 Email:ahmed.shalash@Mohammed NafieCairo UniversityDepartment of Electronics and CommunicationsGeeza,EgyptEmail:mnafie@.egAbstract—There is an ever growing need for higher speed communications over existing wireline plant.To achieve even higher throughput,bonding and joint op-timization of multiple-input-multiple-output cable bundles attracted attention recently.In this paper wefirst explain a technique which allows several users to use the same communication medium while at the same time allowing their transmission to be separated at the receiver.This technique would handle single-channel-multiple-users case. The technique is then extended to handle multiple input multiple output(MIMO)channel by jointly optimizing the transmitfilters to maximize the throughput over the MIMO channel.This optimization is suitable for multiple-channels-single-user and multiple-channels-multiple-users cases.A2-D MIMO channel example is given and sim-ulation results show that the joint optimization achieves higher throughput than independent optimization for each channel.I.I NTRODUCTIONWith the full deployment of wide band digital commu-nications over wireline channels,many new interesting technical as well as marketing problems arise.The problem of optimizing transmitfilters for increasing the throughput of the communication channel has attracted attention lately[1],[2],[3].The research in that area was primarily concerned with single user transmitting over additive white Gaussian noise channels.However,the need for multi-user or multi-task links can be envisioned for homes or small business users.One of these applica-tions is exhibited by the so-called channelized voice over digital subscriber loops(DSL).For these applications, the users can utilize one of the classical multiple access approaches but none are without drawbacks.A transmitfilter optimization technique for the multiple-access environment was shown in[4].However, with the growth of the DSL,the need for even higher data rates keeps mounting.This,coupled with the fact that the cost offiber deployment still makes it attractive to utilize the loop plant to achieve maximum possible throughput,led to the emergence of such project as the G.BOND.The G.BOND project,adopted by the International Telecommunication Union(ITU),supports the topology that one user would combine multiple DSL pairs to increase the data rate.This opens the door for jointly optimizing the transmitfilters to maximize the overall throughput over multiple links.The work done in the dynamic spectrum management project[5] clearly shows the advantage of jointly optimizing the MIMO channel to achieve better data rates.In this paper, numerical optimization is performed on FIRfilter bank to maximize the information rate.In this paper,we introduce an optimization technique that can be utilized to allow either multiple-access over a single wireline channel(single loop)or single/multiple user(s)operating on multiple loops.We model the prob-lem of the multiple loops as a MIMO channel problem. The authors cannot emphasize enough that even though the problem formulation as afilter bank,[6],[7],[8], looks similar to the known multi-dimensional modula-tion used in wireless communications,there are many fundamental differences between wireless and wireline problems.First the wireline channels are linear-time-invariant(LTI).Second,unlike the wireless channels, complete synchronization can be guaranteed for the wireline case.Third,unlike the wireless channel,the wireline channel can be measured and known a priori. Previous work show efficient ways to estimate the chan-nel characteristics[9][10].The receiver is assumed to be a multi-dimensional DFE[11].The remainder of the paper is divided as follows. Section II has the problem formulation.Section III has the solution methodology for fractionally-sampled multi-dimensionalfilter banks over a single channel.SectionIV extends the problem to multiple channels(loops) by formulating it as a MIMO problem.A2-D MIMO problem example is given to highlight the advantage of joint optimization.Section V has concluding remarks.II.P ROBLEM F ORMULATIONSeveral studies have been conducted to optimize the transmitterfilter to successfully mitigate the ISI and noise at the receiver with reasonable complexity[1][3]. From a theoretical stand point,one can achieve the maximum throughput over a given channel by either using infinite length transmitfilters or by sending the data in blocks afterfiltering it by non-stationaryfilters. Attempts had been made tofind optimal practicalfilters. For example,in the discrete multitone(DMT)scheme redundancy is added at the transmitter to map the trans-mitted symbols on orthogonal tones.The DMT can be viewed in light offilter bank theory as in[2].In[12],an optimization technique was introduced to maximize the throughput of noisy ISI channels by passing the white input sequence through an FIR linear transmitfilter.The MMSE-DFE is the assumed receiver as it is a widely used receiver topology.Adopting the standard discrete time representation,the y output of an additive-noise dispersive channel at time step k is given byy k=νm=0h m x k−m+n k,(1)where h m=[h l−1,m···h0,m]t is the m th vector of channel coefficients,assuming an oversampling factor of l,andνis the channel memory.The channel input x at time step k is given byx k=νtn=0p n u k−n,(2)where{p i}v t i=1are the transmitfilter coefficients and {u k}is an assumed white unit-energy transmitted se-quence.Under the assumption of prior knowledge of the channel and noise characteristics,the channel throughput can be maximized.As shown in[1]the channel through-put over a block of N output symbols is given byI=1(N+ν)log2(|I N+ν+H h R−1nn HR xx|),(3)where I N+νis the identity matrix of size N+ν,()h denotes the hermitian transpose,H is the channel matrix whosefirst block-row is equal to the(ν+1)length chan-nel impulse response appended by zeros,and R nn and R xx denote the Toeplitz auto-correlation matrices for the noise and the input sequences,respectively.The channel throughput can be maximized by numerically solving the constrained and nonlinear optimization problem using the sequential quadratic programmingalgorithm[12].stationarynoisex(m) x(m+1) x(m+2).....x(m+N−1)Fig.1.Operation of1-D fractionally sampled FIR transmitfilterIII.M ULTI-USER O PTIMIZATIONA.Fractional OptimizationMore optimization can be achieved when the FIR filter is fractionally sampled instead of being symbol rate sampled.Logically,increasing the sampling rate of the FIR will increase the optimization degrees of freedom, hence allowing increase in the channel throughput.The throughput function can be written as in(3),except that nowR xx=P∗R uu∗P h(4) where P is a matrix corresponding to the fractionally spaced transmitfilter to be optimized,and u is the oversampled input bit sequence.Therefore R uu will be a diagonal matrix,with zeros everywhere except in the positions corresponding to symbols.Figure1shows the operation of the1-D fractionally sampled FIRfilter. As an example,the optimization was performed on the linear channel given byh(z)=0.1+0.3z−1+0.5z−2.(5) The resulting information rates for different SNR are plotted in Figure2.As shown in thefigure,the fractional optimization gives a consistent throughput improvement compared to the popular raised-cosine signal[4].B.Multi-Dimensional SignalingUp until now,most of the research in multiuser communications has focused on wireless channels.Many of the ideas developed for wireless mediums cannot be carried over to wireline communications.An obvious way to see this is that,in wireless communications, each user has their own channel characteristics while in wireline they all share the same wire,and hence the same channel.Second,the wireline channel,prac-tically,is not time-varying and synchronization can be easily achieved between different users.The optimizationFig.2.Throughput for fractionally sampled transmitfiltersFig.3.The advantage of the optimized signaling compared to the time division multiple accesstechnique explained above can be used to generate1-dimensional FIR signal.To be able to accommodate multiple users over the same channel,an increase in the system dimensionality has to be made.We propose two ways for obtaining the multi-dimensional FIRfilters. First,direct optimization with a modification to equa-tion(3)can be performed.Specifically,and since we assume independent usersR xx=i P i R ui u iP h i(6)where P i and R ui u i are,respectively,the transmitfiltermatrix and the autocorrelation function of user i.This technique will outperform the time division multiplexing from information rate view point.Figure3shows the advantage of using the optimization multi-dimensional FIR transmitfilters over the time multiplexed single transmitfilter.The other technique offersflexibility in the signals generation and allows incremental addition of users and easier receiver equalization.We use the frequency response of the FIRfilter as a template and employ the signal generation technique explained in[13] to incrementally generate any required number of sig-nals.The optimization problem is stated mathematically asfindingmin{f i}{max(|F i−R|)}s.t.f i is orthogonal to f j,j<i,where F i is the frequency characteristics of thefilter f i and R is the template being used.Notice that when users join the system incrementally,the original users on the system need only to change their rate such that the total rate is equal to the Nyquist rate of the usable bandwidth of the channel.Thefilters are always the same no matter how many users are using the channel.IV.M ULTI-C HANNEL O PTIMIZATIONThe problem defined in the previous section can be extended to cover operating over MIMO channel.An ex-ample of2-D MIMO channel is shown in Figure4.The channel clearly resembles the known definition of direct connections and far end crosstalk(FEXT)connections associated with DSL environment operating in simplex mode.Only FEXT is considered in this example,how-ever the problem can easily be extended to cover full-duplex communications with near-end crosstalk(NEXT) as well.The throughput as defined in equation(3)can be re-defined to account for the crosstalk portion asI1=1(N+ν1)log2(|I N+ν1+H h1R−1n1n1H1R x1x1|)(7) I2=1(N+ν2)log2(|I N+ν2+H h2R−1n2n2H2R x2x2|),(8)Where R n1n1and R n2n2will each include the external noise term and the cross talk term.Under the assumption of independence among the signal sources and between the signal source and the noise,we can writeR n1n1=R nn+((P2H21)h R u2u2P2H21)(9)R n2n2=R nn+((P1H12)h R u1u1P1H12)(10) where R nn is the external noise auto-correlation,assum-ing same noise is added to both paths,H21and H12are the cross talk channel matrices for both paths,and P1 and P2are the transmitfilter matrices.Fig.4.2-input/2-output MIMO channelA.Optimization FlavorsThere are many possibilities for optimization depend-ing on the particular setup.For multiple users case,we can assume same information rate for all users,i.e.fair-ness among users.This will be similar to the constraint put on the optimization performed for the single-channel-multiple-users case.If the computational complexity is of a concern,we can assume same transmit filter for all users,making it less computationally intensive to perform the optimization.Another setup involves,similar to the G.BOND project,that the user owns all the channels and thus we need to optimize the aggregate information rate.There are other possibilities that can be entertained and are beyond the scope of this paper.The authors intend to discuss these cases in an upcoming paper.Fig.5.The forward channel and the far end crosstalk channelB.Numerical ExampleAssume for simplicity that the direct channel is the same as in equation (5)and the crosstalk channel is given byh (z )=0.1−0.1z −1−.2z −2.(11)The direct and crosstalk channels are shown in Fig-ure 5.The optimization was performed by assuming a symmetric MIMO system (H 1=H 2and H 12=H 21)and by forcing the two filters P 1and P 2to be identical.This guarantees that I 1=I 2and hence can be used to guarantee fairness among different users.Even with such limitations,the jointly optimized filter outperforms the individually optimized filter as shown in Figure 6.The received signal and noise profiles are shown in Figure 7.Fig.6.The joint optimization versus individual optimization for I 1(=I 2in this case)When the optimization is done for I 1+I 2,without con-straining I 1=I 2,even better performance can be achieved as shown in Figure 8.This case can be used,for example,when a single user is using multiple pairs and the only concern is to maximize the total throughput.It is worth noting that for high signal-to-noise ratios,the solution converges to the case where P 1=P 2and I 1=I 2.Note that the information rate in Figure 8is I 1+I 2rather than just I 1as in Figure 6.In Figure 6I 1=I 2by definition.C.Higher DimensionsIt is straight forward to increase the dimensionality of the optimization problem.For practical DSL environ-ment,e.g.a single user owning 2-pair,or 4-pair cable,the MIMO optimization can be performed off line using a central processing unit.Notice that the optimization suitable for each case would depend on the particularFig.7.The PSD profile at the y1point in Figure 4scenario.To give a simple example of that,assume that we want to maximize the throughput of user 1who owns a 2-pair while at the same time keeping fairness between him and user 2with a 4-pair.Then you would maximize the throughput of the user 1while constraining the user 2throughput to be half per channel (or perhaps a little bit less than half)compared to user 1.As the channel is assumed to be practically LTI,then this operation needs to be performed once.For practical systems,temperature changes might contribute to channel variations.However,temperature changes are usually very slow and can be neglected for all practical purposes.Fig.8.The joint optimization of I 1+I 2when P 1and P 2are differentV.C ONCLUSIONIn this paper,the problem of optimizing the FIR trans-mit filter bank was extended from a single channel envi-ronment to MIMO environment.The problem was firstdefined for the single channel,fractionally sampled case.The throughput optimization was performed for multi-dimensional transmit filter bank over a single channel to support single-channel-multi-users case.By re-driving the throughput formula to cover the multi-channel case,the optimization can be jointly performed to maximize the overall information rate for any given MIMO chan-nels.It was shown that doing joint optimization clearly outperforms the independent “stand-alone”optimization.Even though the problem complexity would exponen-tially increase with the number of pairs,for practical DSL bonding environment,entailing two,three,or four-pair bonding,the optimization can be performed with reasonable complexity.Different flavors of optimization can be performed to accommodate different services.These results come in line with the work performed in [5].R EFERENCES[1]N.Al-Dhahir and J.Cioffi,“Block Transmission Over Disper-sive Channels:Transmit Filter Optimization and Realization,and MMSE-DFE Receiver Performance,”IEEE Trans.on Infor.Theory ,pp.137–160,Jan.1996.[2] A.Scaglione ,S.Barbarossa,and 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