Solutions Manual for Communication Systems 4th Edition Simon Haykin McMaster University, CanadaPreface This Manual is written to accompany the fourth edition of my book on Communication Systems. It consists of the following: * Detailed solutions to all the problems in Chapters 1 to 10 of the book + MATLAB codes and representative results for the computer experiments in Chapters 1, 2, 3, 4, 6, 7,9 and 10 I would like to express my thanks to my graduate student, Mathini Sellathurai, for her help in solving some of the problems and writing the above-mentioned MATLAB codes. I am also grateful to my Technical coordinator, Lola Brooks for typing the solutions to new problems and preparing the manuscript for the Manual. Simon Haykin Ancaster April 29, 2000CHAPTER I roblem As an illustration, three particular sample functions of the random process X(t), corresponding to F = W/4, W/2, and W, are plotted below: sin (2nWe) t arty sin(an # t inton & sin(an ft) t -2 ° 2 Ww Ww To show that X(t) is nonstationary, we need only observe that every waveform illustrated above is zero at t = 0, positive for 0 < t < 1/M, and negative for -1/2W < t < 0. ‘Thus, the probability density function of the random variable X(t,) obtained by sampling X(t) at t, = 1/tW is identically zero for negative argument, whereas the probability density function of the random variable X(t) obtained by sampling X(t) at t = -1/IW is nonzero only for negative arguments. Clearly, therefore, fe? # fee) "D> and the random process X(t) is nonstationary.Problem 1.2 X(t) = A cos(2nf,t) efore, X, = A cos(2nf,t,) Since the amplitude A is uniformly distributed, we may write 1 waar 0 < ¥, < costar t,) fy (x) = cra °, otherwise Fx er cos (2mEt, = 4 ° cos (2r£_t,) Similarly, we may write Xi, =A coslant (ty) and 1 {aaah O ) = costam(t -tp)I >) of the procs Since every weighted sum of the samples, Z(t) is Gaussian, it follows that Z(t) is a Gaussian process. Furthermore, we note that 2 2 eae) 7 EC = 1 This result is obtained by putting t,=t, in Eq. (1). Similarly, 2 2 E(Z2 e P28) * ElZ*(t,)] = 1 Therefore, the correlation coefficient of Z(t,) and Z(tp) is Covl2( t4)20 ty I @ 2 4)72(ty) Hence, the joint probability density function of Z(t,) and 2(tp) Fo¢ 4) 52g) BrP Be) = © exl-RC2 4422) where 1 awh ~cos"2e (t=t9)] 1 = Es By sintan(e,-ta) Q(24425) 2 sin“L2n(ty-to)](>) We note that the covariance of Z(t,) and Z(t) depends only on the time difference tyrtg. The process Z(t) is therefore widessense stationary. Since it is Gaussian it is also strictly stationary. Problem 1.5 @) Let X(t) = A + ¥(t) where A is a constant and Y(t) is a zero-mean random process. The autocorrelation function of X(t) is * Ry(e) = ELK(ter) X(t)] E((A + ¥(ter)] [A + ¥(t)]} Eta? + A Y(ter) +A Y(t) + Y(ter) Y(t) a RCD Which shows that Ry(t) contains a constant component equal to A°. (by Let X(t) = Ay cos(2uf .t + 6) + Z(t) where Ay cos(2nf .t+é) represents the sinusoidal component of X(t) and @ is a random phase variable. The autocorrelation function of X(t) is R(t) = E(K(ter) X(t)I EUIA, cos(2nft + 2nfjx + 6) + Z(t+r)] (A, cos(2nft + 8) + Z(t)]} ELAZ cos(2af,t + 2nfyr + 6) cos(2ift + 8)1 + ElZ(ter) Ay cos(2nf + 6)] + ELA, cos (anf t + ant. + 6) 2(t)] + EZ (ter) 2¢t)) 2 (Ag/2) cost2nt.t) + Ry(x) which shows that Ry(t) contains a sinusoidal component of the same frequency as X(t). Problem 1.6 (a) We note that the distribution function of X(t) isoO, x <0 Ad rAd Ace Faces® and the corresponding probability density function is 1 1 f(g) = FOX) + 3 Cx - A) which are illustrated below: a oe 1.0 (b) By ensemble-averaging, we have EIX(t)] = Sx fycgy (2) ax £ x Oh 6G) +b oex - a0) ax A - 2 The autocorrelation function of X(t) is Ry(t) = E(X(ter) X¢t)] Define the square function Sqq (t) as the square-wave shown below: 0Sa, (t) 1.0 Then, we may write Ryle) = EIA Sag Ct - 1 Say (t= ty +t) Say (t= ty) + goat, 5 ‘a Ty a * Ta 7 4 ea 24h vic 2. 2 es 2 Since the wave is periodic with period Tg, Ry(t) must also be periodic with period Ty. (c) On a time-aver aging basis, we note by inspection of Fig. P/.4that the mean is ext = £ Next, the autocorrelation function Ty/2 a x(ter) x(t) dt Tye L 1 Kx(ter)x(t)> = has its maximum value of A°/2 at t = 0, and decreases linearly to zero at t = Tg/2. Therefore, 2 emcees éxtter) x(t)? #5 = 2 0Again, the autocorrelation must be periodic with period Ty. (4) We note that the ensemble-aver aging and time-averaging procedures yield the same set of results for the mean and autocorrelation functions. Therefore, X(t) is ergodic in both the mean and the autocorrelation function. Since ergodicity implies wide-sense stationarity, it follows that X(t) must be wide-sense stationary. Problem 1.7 (a) For tr] > T, the random variables X(t) and X(ter) occur in different pulse intervals and are therefore independent. Thus, E(X(t) X(ter)) = E(X(t)] E(K(ter)], lot Since both amplitudes are equally likely, we have E(X(t)] = Elx(ter)] for It] > T, 4/2, Therefore, 2 A Rt) =e For It] <1, the random variables occur in the same pulse interval if t, s ) Form the non-negative quantity EUUXCtar) + ¥(t)}7) = ELK? (ter) + ax(ter) Y(t) + ¥2(t)) E(x*(ter)] + 26(x(ter) ¥(t)) + ELY@(t] RCO) + ARyy(t) + RyCO) Hence , Ry(O) + Aye) + Ry(O) > 0 or Wy 1 ) Since Ryy(t) = Ryy(-t), we haveRyy(t) =F nCu) Ry(-r-u) du Since Ry(t) is an even function of t: Ryy(t) = nCu) Ry(t+u) du Replacing u by -u: Rye) =F now) Ry(e-u) du (c) If X(t) is a white noise process with zero mean and power spectral density No/2, we may write z Rye) = 260) Therefore, Rye) = ges aw) 6-u) du Using the sifting property of the delta function: N ‘0 Ryy(t) 2 xe hte) That is, 2 h(t) = Ge Ryy (1) Ny “yx This means that we may measure the impulse response of the filter by applying a white Power noise of, spectral density No/2 to the filter input, cross-correlating the filter output with the input, and then multiplying the result by 2M. Problem 1.12 (a) The power spectral density consists of two components: (1) A delta function 6(t) at the origin, whose inverse Fourier transform is one. (2) A triangular component of unit amplitude and width 2 oq, centered at the origin; the inverse Fourier transform of this component is fy sine@(f gt). Therefore, the autocorrelation function of X(t) is 13Ryle) = 1 + fq sine@(t ye) Wnieh is sketched below: R(t) T 1 1 i fy L fo (b) Since Ry(t) contains a constant component of amplitude 1, it follows that the de power contained in X(t) is 1. (c) The mean-square value of X(t) is given by EIX?(t)] = Ry (0) F14fy The ac power contained in X(f) is therefore equal to fo (4) If the sampling rate is f>/n, where n is an integer, the samples are uncorrelated. They are not, however, statistically independent. ‘They would be statistically independent if X(t) were a Gaussian process. ler The autocorrelation function of no(t) is + By Ce getg) = Elnatty) nat] E{fn (ty) cos(anfgt, +0) = ny(t,) sin(2nt.t40)] + [nj(tp) cos(arfgt, +0) - nyt.) sin(arf.t, + 0)]) Eln,(t,) nj(ty) cos(anf.t, +6) cos(anr.t, + 6) = nylt,) ny(ty) cos(arf,t, +0) sin(arf.t, + @) = nytty) ny(tp) sin(2nt.t, +6) cos(arf.t, +6) 14+ ny(ty) nylty) sin(ant ty +8) sin(art.t, +6)) = E{ny(ty) ny(to) coslarty(t yt.) = ny(ty) ny(tg) sinlant, (tty) + 26)) = Elny(ty) ny(to)] coslart,(t j-t3)1 ~ Elny(t,) ny(ty)] + Elsinfant,(t +t.) + 20]} Since © is a uniformly distributed random variable, the second term is zero, giving Ry (tyta) = Ry (tyty) coslant(t to] 2 1 2) Since n,(t) is stationary, we find that in terms of t tyrtet Ry @) Ne By cos (af ,t) Taking the Fourier transforms of both sides of this relation: Sy (f) = 15 (fet) + Sy, (F893 1 With Sy (f) as defined in Fig. Pfy% we find that Sy (f) is as show below: 1 2 20 20 MpaProblem 1.14 The power spectral density of the random telegraph wave is S(t) =F Ryle) expl-Jertt) a o =f exp(2vt) exp(-janft) de +f expl-2ur) exp(-Jjarfr) dr 0 0 1 = Spape) leepl@vr - Senfe)) 1 ~ Bwaget) Lexar - gered) 5 © 20=5nf) * 204jnt) van The transfer function of the filter is HOD) = 79a T+ jarf rc Therefore, the power spectral density of the filter output is 2 SyCf) = THCA I? scr) [1 + (nfRC)?1 an) To determine the autocorrelation function of the filter output, we first expand Sy(f) in partial fractions as follows v 1 — a - . Taner? Cyan) ete ve Sy(f) = Recognizing that 15v 2 yee exp(-2t tt) > v2 snr 172Rc 2, 2pe exp(-1t{ /RC) => (1 /2RC) “ag We obtain the desired result: L expe In) = 2Rc exp #1) Ryle) = Kr 5 1-4R 70% S 16Problem 1.15 We are given y= fsa For x(t) = 6(2), the impulse response of this running integrator is, by definition, a(n =f 8(t)de 1 =1 for 1-TSOSt or, equivalently, 0S1 Next, we note that Ry(x) is related to the power spectral density by Ry) = f Sy(f) cos(anft) df (2) power Therefore, comparing Eqs. (1) and (2), we deduce that the, spectral density of X(t) is When the frequency assumes a constant value, f, (say), we have e a1 1 fpf) = 5 b(t.) + 3 6( fet) Ci 19and, correspondingly, 2 2 A A Sy() = FF Olof.) + ster) Problem L18 Let 6x? denote the variance of the random variable X, obtained by observing the random process X(t) at time t,. The variance ,? is related to the mean-square value of X,, as follows ox = IX?) - ne wherefx = EX]. Since the process X(t) has zero mean, it follows that of = EX?) Next we note that EIX{] = [7 Sx(Oaf We may therefore define the variance ox” as the total area under the power spectral density Sx(f) as oe £ Sx(Oat @ ‘Thus with the mean px = 0 and the variance o, defined by Eq. (1), we may express the probability density function of X, as follows fx) = me oian ox 20x 20Problem 1.19 The input-output relation of a full-wave rectifier is defined by { X(tyy— X(ty) 20 [xii xt) <0 Y(t) = XC Y= Tne probability density function of the random variable X(t,), obtained by observing the input random process at time t,, is defined by To find the probability density function of the random variable Y(t,), obtained by observing the output random process, we need an expression for the inverse relation defining X(t.) in terms of Y(t,). We note that a given value of Y(t,) corresponds to 2 values of X(t,), of equal magnitude and opposite sign. We may therefore write X(ty) = KCI, KC) <0 XE, = Wb, Ky > 0. In both cases, we have - px | aes) eae ‘The probability density function of Y(t,) is therefore given by ! X(t, ) | Bore fy OY) = fyq s(x = oy) [ae]. (ezys X(t) X(t) art) XC) j He,which is illustrated below: fa) Ye) 0.791 Problem 1.20, (a) Tne probability density function of the random variable Y(t,)+ obtained by observing the rectifier output Y(t) at time t,, is (2 1 exp(- dy 20 (z oy cc aa f, (y) = X(t) oOo, y, is given by by) = J X(tga) hy(u) du The expected value of Z(t») is my, * Elatt,)1 HAO) my where Hg(0) =F pCu) au The covariance of Y(t.) and Ut) is Cove) Zby)] = EL((ty) ay MZ(ty) ~My II 1 2 ELL Slt yt) Ay) (tye) = Ay) by) hy Cu) dt du) Lf ELK Ht) ay) K(tgeu) = 44) 46D hy(u) dt du LF Cy(tyatgeteu) ny Cr) hy(u) at du mee X al 1 2! where Cy(t) is the autocovariance function of X(t). Next, we note that the variance of X(t.) is 2 2 Of = LCE) may 97 FEF Sylora) yD Cw) de du and the variance of Z(t) is 2 2. oy = EL(Z(ty) — a, °7 2, 2 25 es fete) h(t) ho(u) dt du 26The correlation coefficient of ¥(t4) and Z(t) is covi¥(t4)2(t5)1 3 YT, % pe Since X(t) 18 a Gaussian process, it follows that Y(t,) and 2(t,) are 4 i a : ointly Caussi with a probability density function given by ' a sonny fewseten FyC4 4) ,2¢ty) Ya22) = K expl-O0y92)1 where anoy a 41 1 | ein 2 — »? - 206 2a-e)| ¥yn1. 227 z, 171 22, a 22 2,2 Y. 2. 7. 2 2 Ay 9z5) = 1 1 (b) The random variables ¥(t,) and Z(t,) are uncorrelated if and only if their covariance is zero, Since ¥(t) and Z(t) are jointly Gaussian processes, it follows that Y(t,) and 2p) are statistically independent if Cov[¥(t,)2(tp)] is zero. ‘Therefore, the necessary and sufficient condition for ¥(t,) and Z(t,) to be statistically independent is that is £ Sgt grtgeteu) h(a) hg(u) dt du = 0 for choices of t, and tp. .Problem 1.22 (a) The filter output is Y(t) = f h(x) X(tet) de a Tf Xtar) dt 0 + Then, the sample value of Y(t) at t=T equals ie 5 Yaqs xt) du ° ‘The mean of ¥ is therefore T EW] = EES x(u) au) oO T J E(XK(u)] du 0 aio The variance of Y is of = ELV?) - terry}? = Ry(0) = ce Sy(f) df £ S(t) ce? atH(f) = s W(t) exp(—Jj2nft) dt T J expl-jextt) dt 0 sI- pexptoseert 7 =jext 4 0 = jar (1 = exp(-j2fT)] = sine(fT) exp(-jafT) Therefore, og = S SyCt) sine@(eT) at (b) Since the filter input is Gaussian, it follows that Y is also Gaussian. probability density function of ¥ is mere of is detined shove. Problem 1.23 G@) Te power spectral density of the noise at the filter output is given by N = Moy jee 2 Se rorya Hence, they 2 sy(t) = 30 2xflmy T+(2qfL/RY zu Ls 14(2qf L/R)' Zz J ‘The autocorrelation function of the filter output is therefore x Bo too - Be expe Bet) ‘The mean of the filter output is equal to H(0) times the mean of the filter input. The process at the filter input has zero mean. The value H(0) of the filter’s transfer function H(f) is zero. It follows therefore that the filter output also has a zero mean. ‘The mean-square value of the filter output is equal to Ry(0). With zero mean, it follows therefore that the variance of the filter output is of = RCO) Since Ry(t) contains a delta function &(t) centered on t = 0, we find that, in theory, oy? is infinitely large. 30Problem 1.24 (a) The noise equivalent bandwidth is 4s wey ae (0)\? +» ipa ne + (4/8) at zs af ors (ety) fo * Bn sinGa/en) 2 * sino(i 72a) (b) When the filter order n approaches infinity, we have 0 Me SrRStT ERT Problem 1.25 ‘The process X(t) defined by xX®= Yo hit - wy, kee where h(t - %,) is a current pulse at time %, is stationary for the following simple reason. There is no distinguishing origin of time. 31Problem 1.26 (a) Let S,(f) denote the power spectral density of the noise at the first filter output. The dependence of S,(f) on frequency is illustrated below: s\(2) ¥/2 Let S,(f) denote the power spectral density of the noise at the mixer output. may write So(f) 1 TUS (fe) + 84 Ce which {s illustrated below: 8, (£) 32 Then, we‘The power spectral density of the noise n(t) at the second filter output is therefore defined by No ee 860 = 4 uc 0, otherwise ‘The autocorrelation function of the noise n(t) is RO = nce sine(2Bt) (b) The mean value of the noise at the system output is zero. Hence, the variance and mean-square value of this noise are the same. Now, the total area under S,(f is equal to (No/4)(2B) = NoB/2. The variance of the noise at the system output is therefore NyB/2. (c) The maximum rate at which n(t) can be sampled for the resulting samples to be uncorrelated is 2B samples per second, 33coblem 1.2’ (a) The autocorrelation function of the filter output is Ryle) = FF ley) Wlrg) Ryltoty+ty) dey Aty i} Since Ry(t) = (No/2) 6(t), we find that the impulse response h(t) of the filter must Satisfy the condition: Ry(r) = zh i h(t,) h(t.) S(tHt 445) dt, dty Ny * Bf Mart) h(ty) dtp (b) For the filter output to have a power spectral density equal to Sy(f), we have to choose the transfer function H(f) of the filter such that Ng 2 Sy(f) = so HCE) or (aCe)Problem 1.28 (@) Consider the part of the analyzer in Fig. 1.19 defining the in-phase component 742), reproduced here as Fig. 1 Narrowband noise no) filter > ni) 2cos(2nf.t) Figure | For the multiplier output, we have v(t) = 2n(t)cos(2mf,t) Applying Eq. (1.55) in the textbook, we therefore get SVP) = SF fo) +50 F + fod] Passing v(#) through an ideal low-pass filter of bandwidth B, defined as one-half the bandwidth of the narrowband noise n(f), we obtain Sy) = | Sy(f) for -BS f0 oe 20° ~ far) = | (oo otherwise To evaluate the variance 0°, we note that the power spectral density of n (t) or n_(t) is as follows zr Q 39Sy (fs, (£) Tr q Since the mean of n(t) is zero, we find that 2 os 2NB ‘Therefore, ROM = (op. Tue,mean value of the envelope is equal to VaNGB, and its variance is equal to «858 NOB. 0Problem 1.32 Autocorrelation of a Sinusoidal Wave Plus White Gaussian Noise In this computer experiment, we study the statistical characterization of a random process X(t) consisting of a sinusoidal wave component Acos(2nf.t + @) and a white Gaussian noise process W() of zero mean and power spectral density No/2. That is, we have X(1) = Acos2nf,t+@+ W(r) 0) where © is a uniformly distributed random variable over the interval(-at,1). Clearly, the two components of the process X(t) are independent. The autocorrelation function of X(0) is therefore the sum of the individual autocorrelation functions of the signal (sinusoidal wave) component and the noise component, as shown by 2 N Ry(t) = Acos(any,2) +528(t) This equation shows that for |t] > 0, the autocorrelation function Ry(t) has the same sinusoidal waveform as the signal component. We may generalize this result by stating that the presence of a periodic signal component corrupted by additive white noise can be detected by computing the autocorrelation function of the composite process X(t). ‘The purpose of the experiment described here is to perform this computation using two different methods: (a) ensemble averaging, and (b) time averaging. The signal of interest consists of a sinusoidal signal of frequency f, = 0.002 and phase 0 = - 1/2, truncated to a finite duration T = 1000; the amplitude A of the sinusoidal signal is set to V2 to give unit average power. A particular realization x(¢) of the random process X(2) consists of this sinusoidal signal and additive white Gaussian noise; the power spectral density of the noise for this realization is (No/2) = 1000. The original sinusoidal is barely recognizable in x(0), (a) For ensemble-average computation of the autocorrelation function, we may proceed as follows: + Compute the product x(t + t)x(¢) for some fixed time t and specified time shift t, where x(t) is a particular realization of the random process X(#). + Repeat the computation of the product x(t + t)x(1) for M independent realizations (i.c., sample functions) of the random process X(t). + Compute the average of these computations over M. + Repeat this sequence of computations for different values of t. The results of this computation are plotted in Fig. 1 for M = SO realizations. The picture portrayed here is in perfect agreement with theory defined by Eq. (2). The important point to note here is that the ensemble-averaging process yields a clean estimate of the true 4autocorrelation function Ry(t) of the random process X(t). Moreover, the presence of the sinusoidal signal is clearly visible in the plot of R(t) versus t. (b) For the time-average estimation of the autocorrelation function of the process X(t), we invoke ergodicity and use the formula Rea) = lim R(t, 7) @) where R,(t,7) is the time-averaged autocorrelation function "x(t t)xttyar (4) 1 ‘The x(1) in Eq. (4) is a particular realization of the process X(1), and 27 is the total observation interval. Define the time-windowed function ap(t) = [0 -TSIST 5 mh 0, otherwise ©) We may then rewrite Eq, (4) as nl RUG T) = spf arle4 t)ap(tpat © For a specified time shift t, we may compute R,(t,7) directly using Eq. (6). However, from a computational viewpoint, it is more efficient to use an indirect method based on Fourier transformation. First, we note From Eq. (6) that the time-averaged autocorrelation function R,(t,T) may be viewed as a scaled form of convolution in the t-domain as follows: RAT) Af ao x(t) ) where the star denotes convolution and x/(t) is simply the time-windowed function x7(#) with replaced by 7. Let X7(f) denote the Fourier transform x7(t); note that X;(/) is the same as the Fourier transform X(f,7). Since convolution in the t-domain is transformed into multiplication in the frequency domain, we have the Fourier-transform pair: RT) = Ef [xP ®The parameter |X7()|?/27 is recognized as the periodogram of the process X(t). Equation (8) is a mathematical description of the correlation theorem, which may be formally stated as follows: The time-averaged autocorrelation function of a sample function pertaining to a random process and its periodogram, based on that sample function, constitute a Fourier- transform pair. We are now ready to describe the indirect method for computing the time-averaged autocorrelation function Ry(t,7) + Compute the Fourier transform X7(f) of time-windowed function x7(1). + Compute the periodogram |X7(f/?/2T. + Compute the inverse Fourier transform of |X()/?/27- To perform these calculations on a digital computer, the customary procedure is to use the fast Fourier transform (FFT) algorithm, With x(t) uniformly sampled, the computational procedure described herein yields the desired values of | R,(t,7)_ for 1 = 0,4,2A,-,(N-1)A where A is the sampling period and N is the total number of samples used in the computation. Figure 2 presents the results obtained in the time-averaging approach of “estimating” the autocorrelation function Ry(t) using the indirect method for the same set of parameters as those used for the ensemble-averaged results of Fig. 1. The symbol Rx(t) is used to emphasize the fact that the computation described here results in an “estimate” of the autocorrelation function Ry(t). The results presented in Fig. 2 are for a signal-to-noise ratio of + 10dB, which is defined by A AT No/QT) No (9) On the basis of the results presented in Figures 1 and 2 we may make the following observations: + The ensemble-averaging and time-averaging approaches yield similar results for the autocorrelation function Ry(t), signifying the fact that the random process X(¢) described herein is indeed ergodic. + The indirect time-averaging approach, based on the FFT algorithm, provides an efficient method for the estimation of Ry(t) using a digital computer. + As the SNR is increased, the numerical accuracy of the estimation is improved, which is intuitively satisfying, 431 Problem 1.32 Matlab codes 4% Problem 1.32a CS: Haykin 4% Ensemble average autocorrelation %M. Sellathurai clear all Arsqrt (2); % signal s=cos(2epitf_cxt); Ynoise snr = 10°(SNR4b/10); ‘randn(1, Length(s) ))/sqrt(snr)/sqrt(2); gna plus noise autocorrelation [e_corrs]=en_corr(s,s, N} averaged autocorrelation -corrt_tte_corrt} wprints plot (~600:500-t,¢_corrt_t/M); xlabel(?(\tau)?) ylabel R_K(\tau) ") 44% Problem 1.32b 0S: Haykin 4 time-averaged estimation of autocorrelation %M. Seliethurai clear all % signal s=cos(24piet_cet) :Ynoise Ynoise snr = 10°(SNRdb/10); wn = (randn(1,2length(s)))/sqrt (snr)/eqrt(2); Ysignal plus noise sestwn; 4% time -averaged autocorrelation [e.corrf]=time_corr(s,N); Yprints plot (~500:500-1,e_corrt); xlabel(?(\tau)') ylabel (?R_X(\tau)’) 45function [eorr#} % ensemble average 4% used in problem 1.32, cS: Haykin %M. Sellathurai, 10 june 1999. ncorr(s, v, NY funtion to compute the autocorreation/ crose-correlatic nax_cross_cors teslength(u); tor shitted_u=(u(mti:tt) a(i:m)]; corr (m+1)=(sum(v.*shitved_u))/(N/2); if (abs (corr)>max_cross_corr) max_cross_corr=abs(corr); end end 46function [corrf]=time_corr(s,N) 4% tuntion to compute the autocorreation/ cross-correlation % time average 4% used in problem 1.32, CS: Haykin %M. Sellathurai, 10 june 1999, (5); Aa=tttshift ((abs(X).°2)/(N/2)); corrf=(tftshift (abs (ifft(X1)))); 47Answer to Problem 1.32 Figure 1: Ensemble averaging eas aa! ats mats — Mats ap ate ates — ats its ab Figure 2: Time averaging, 48Problem 1.33 Matlab codes % Problem 1.33 CS: Haykin % multipath channel %M. Sellethurai clear all 5% initializing counters 10000; % number of samples act; % line of sight component component for i=1:P Assqrt (randn(W,M).°2 + randn(N,M).-2); +cos(cos(rand(N,M)*2epi) + rand(N,W)*24pi); % inphase epmponent -#8in(cos(rand(w,M)*2epi) + rand(N,")*2*pi); quadrature phase component sum(xi?)); ssum(xq’))} x¥ rarsqrt((xita).~2+ xq.°2) ; % rayleigh, rician fading Gh X]=hist (ra,60); % print plot (Kt ,Nf/(sum(Xf .#Nt)/20)) xlabel(’v’) ylabel(*£_v(v)*) 49Answer to Problem 1.33 Figure 1 Rician distribution 50Problem 2, (a) Let the input voltage v; consist of a sinusoidal wave of frequency $f, (i.e., half the desired carrier frequency) and the message signal m(t): vy = A, cos(nf t)+m(t) Then, the output current 1, is : 3 tot ayy tag vy aj{A,cos(af,t)+n(t) ]easth costa ft)+m(t)]> ay(Ajcostnt t)mt)] + Jagh3 [eos(ief,t)+3eosnf,t)] + Bagt2 w(erttecostantgt)] + 3ahjcostnt tart) + age) Assume that m(t) occupies the frequency interval -W 20 a . 20 4 aa SLFrom this diagran we see that in order to extract a DSBSC wave, with carrier frequency f, from i,, we need a band-pass filter with mid-band frequency f, and bandwidth 2, which satisfy the requirement: f fw >See that is, f, > Gt Therefore, to use the given nonlinear device as a product modulator, we may use the following configuration: | device BEE ° 3. 42 7 43 Al m(t) cos (2ne_t) A cos (net) m(t) (b) To generate an AM wave with carrier frequency f, we require a sinusoidal component of frequency £, to be added to the DSBSC generated in the manner described above. To achieve this requirement, we may use the following configuration involving a pair of the nonlinear devices and a pair of identical band-pass filters. Nonlinear BPF device A cos (7) &) AM wave Nonlinear BPF device 52The resulting AM wave is therefore $a, AaCAgem(t)leos(2rf,t). Thus, the choice of the de level Ag at the input of the lower branch controls the percentage modulation of the AM wave. Problem 2.2 Consider the square-law characteristic: volt) = ayvy(t) + agvzit) a where a, and a, are constants. Let v(t) = Agcos(2nfgt) + m(t) @ ‘Therefore substitutingEq. (2) into (1), and expanding terms: volt) = ayAj1 + 222 mit) | cos(2nfyt) LOR : @) + aym(t) + apm %(t) + agA2cos*(2nf,t) ‘The first term in Eq. (3) is the desired AM signal with k, = 2ay/a,. The remaining three terms are unwanted terms that are removed by filtering. Let the modulating wave m(t) be limited to the band -W < f< W, as in Fig. 1(a). Then, from Eq. (3) we find that the amplitude spectrum |V,(f)| is as shown in Fig. 1(b). It follows therefore that the unwanted terms may be removed from vz(t) by designing the tuned filter at the modulator output of Fig. P2.2 to have a mid-band frequency f, and bandwidth 2W, which satisfy the requirement that f, > 3W. Wvs00 won KA A A A WO WF 2h “haw WOW Wh mh he (@) (b) aw Figure 1 33Problem 2.3 The generation of an AM wave may be accomplished using various devices; here we describe one such device called a switching modulator. Details of this modulator are shown in Fig. P2.3a, where it is assumed that the carrier wave c(t) applied to the diode is large in amplitude, so that it swings right across the characteristic curve of the diode. We assume that the diode acts as an ideal switch, that is, it presents zero impedance when it is forward-biased [corresponding to c(t) > 0}. We may thus approximate the transfer characteristic of the diode-load resistor combination by a piecewise-linear characteristic, as shown in Fig. P2.3b. Accordingly, for an input voltage v(t) consisting of the sum of the carrier and the message signal: v,(0) = A,cos(2n ft) +m(t) a) where |m(2)| << A,, the resulting load voltage v9(0) is, v(t), o(t)>0 0, e(t)<0 vl) = @ ‘That is, the load voltage v(t) varies periodically between the values v;(1) and zero at a rate equal to the carrier frequency /,. In this way, by assuming a modulating wave that is weak compared with the carrier wave, we have effectively replaced the nonlinear behavior of the diode by an approximately equivalent piecewise-linear time-varying operation, We may express Eq. (2) mathematically as v(t) = A, cos(2nf1) + m(Ngr,(0) (3) where g7,(#) is a periodic pulse train of duty cycle equal to one-half, and period Ty = Mf, as in Fig. I. Representing this g,(#) by its Fourier series, we have 1,2 (—1 bri = 542 cos[2mf,t(2n-1)] 4) ‘Therefore, substituting Eq, (4) in (3), we find that the load voltage v2(7) consists of the sum of two components: 1. The component 1m] 0s (27, f,.t) 34which is the desired AM wave with amplitude sensitivity k, = 4m4,. The switching modulator is therefore made more sensitive by reducing the carrier amplitude A,; however, it must be ‘maintained large enough to make the diode act like an ideal switch. 2. Unwanted components, the spectrum of which contains delta functions at 0, +2/. +4fe. and so on, and which occupy frequency intervals of width 2W centered at 0, 43f,.. 45f,, and so on, where W is the message bandwidth. aryl i To te : why wt Fig, |: Periodic pulse train The unwanted terms are removed from the load voltage v2(*) by means of a band-pass filter with mid-band frequency f, and bandwidth 2W, provided that f, > 2W. This latter condition ensures that the frequency separations between the desired AM wave the unwanted components are large enough for the band-pass filter to suppress the unwanted components.(a) The envelope detector output is v(t) = All1+ woos (anf t) | which is illustrated below for the case when p= We see that v(t) is periodic with a period equal to f,, and an even function of t, and so we may express v(t) in the form: v(t) = ag +2 Ea, cos(2nn t,t) nl Vet, art vit)dt mo where 13 q 1fy = At s (142 cos(arf tyidt + 2hf. s (-1-2008(arf,t)1dt Gh, ‘ f) orm ae m ™ © sin(E am Vet, = ats v(t) cos(2nm ft) dt ny im 1438, # Abeta J (142c0s(2nf,,t) Jeos(2nr ft) dt 561peE, n + Att (-1-2cos(2rf,.t) Jeos(2nx ft) dt 1/3Ey 2 29 ¢2 otnc22%) = atntond] + eS (2 stnt2EKowty] = stats (mt 91) = FB 2 sin@% ~ sine + ate En te {2 int 2 1)] sin(n (n=1)]) (2) + Gete 2 sinl2h(n-1)) ~ sinta (ne For nz0, Eq. (2) reduces to that shown in Eq. (1). (b) For n=1, Eq. (2) yields Gs} an * D For n=2, it yields 1 an Therefore, the ratio of second-harmonic amplitude to fundamental amplitude in v(t) is 2 38 = 0.452 By nSProblem 2.5 (a) The demodulation of an AM wave can be accomplished using various devices; here, we describe a simple and yet highly effective device known as the envelope detector. Some version of this demodulator is used in almost all commercial AM radio receivers. For it to function properly, however, the AM wave has to be narrow-band, which requires that the carrier frequency be large compared to the message bandwidth, Moreover, the percentage ‘modulation must be less than 100 percent. An envelope detector of the series type is shown in Fig. P2.5, which consists of a diode and a resistor-capacitor (RC) filter. The operation of this envelope detector is as follows. On a positive half-cycle of the input signal, the diode is forward-biased and the capacitor C charges up rapidly to the peak value of the input signal. When the input signal falls below this value, the diode becomes reverse-biased and the capacitor C discharges slowly through the load resistor R). The discharging process continues until the next positive half-cycle. When the input signal becomes greater than the voltage across the capacitor, the diode conducts again and the process is repeated. We assume that the diode is ideal, presenting resistance r; to current flow in the forward-biased region and infinite resistance in the reverse-biased region. We further assume that the AM wave applied to the envelope detector is supplied by a voltage source of internal impedance R,. The charging time constant (rr + R,) C must be short compared with the carrier period I/f, that is (ry + RICA a) so that the capacitor C charges rapidly and thereby follows the applied voltage up to the positive peak when the diode is conducting. (b) The discharging time constant R;C must be long enough to ensure that the capacitor discharges slowly through the load resistor R; between positive peaks of the carrier wave, but not so long that the capacitor voltage will not discharge at the maximum rate of change of the modulating wave, that is 1 1 FeRCeg Q) where W is the message bandwidth, The result is that the capacitor voltage or detector output, is nearly the same as the envelope of the AM wave. 58vy(t) = Ag1 + kgmi(t)]cos(2nf,t) (2) Then the output of the square-law device is va(t) = ayvy(t) + agv f(t) = aA{1 + kgm(t)]eos(2nf,t) + peokzt + 2kymi(t) + k2m%t)] [1 + cos(4nf.t)] (b) ‘The desired signal, namely ajA,k,m(t), is due to the av} (t) - hence, the name "square-law detection”. This component can be extracted by means of a low-pass filter. This is not the only contribution within the baseband spectrum, because the term 1/2 a)A,"k,”m%(t) will give rise to a plurality of similar frequency components. The ratio of wanted signal to distortion is 2/k,m(t). To sake this ratio large, the percentage modulation, that i, Tiym(F should be kept small compared with unity.Problem Tne squarer output is arg 2 cos? vy(t) = AS Cokgmt)]? cos? (are, t) 1% Tsk m(t) +n? Ct) IE eeosCHet st) 1 The amplitude spectrum of v,(t) is therefore as follows fi the interval Wife: |! + assuming that m(t) is limited to vce] Since f, > 21, we find that 2f,-2W > 2. ‘Therefore, by choosing the cutoff frequency of the low-pass filter greater than 2W, but less than 2f,-2H, we obtain the output vat) Cakegm(t) 7? le’ Hence, the square-rooter output is a c v(t) = 7 Ciakegnte)) de whieh, excent for the de component — , 1s proportional to the message signal a(t). 2 Problem 2.8 (a) For f, = 1.25 klz, the spectra of the message signal m(t), the product modulator output s(t), and the coherent detector output v(t) are as follows, respectively: ©vee) KK (ate) = 0 Fy (bv) For the case when f, = 0.75, the respective spectra are as follows: M(f) - £ (kHz) s(t) = 5° mt) cos(2nart) Thus the output signal is the message signal modulated by a sinusoid of frequency Af. (b) If m(t) = cos(2nr.t), A then sa(t) = 5° cos(2mf it) cos(2natt) 4 Ah obl 2, (a) y(t) = s(t = A? cos®(2at yar (t) 2 = 5f [4cos( aft) Im? (t) 2 ce Therefore, the spectrum of the multiplier output is 2 eS) az ow o £ MOONEE adda + ge LF MOUMCE2E adda +S MOOMCE42f,-2)da) Y(f) = “o’ where M(f) = Fmt) J. (b) at 8 fy We haveae iw =e Yer) #t FMOOM(2E,~ Maa a © © + UL NOOM- da eS MOONCHE apd A] Le a e Since M(=A) = M#(A), we may write a2 c YQE) = BS MONM(2F,-A)ad. LINGO aa ef MOONCE,~ aan) a 2 et F + With m(t) limited to -W < f < Wand f, > W, we find that the first and third integrals reduce to zero, and so we may simplify Eq. (1) as follows 2 Ao oe 2 yer sais imcofPaa where Eis the signal energy (by Rayleigh's energy theoren). Similarly, we find that > Y(-2f,) = GE The band-pass filter output, in the frequency domain, is therefore defined by 2 A Vif) = GPE afl e(e2t.) + 6(fe2r,)) Hence, > 7 v(t) le E af cos(4xt,t) oyProblem 2.12 The multiplexed signal is s(t) = A my(t) cos(2mf.t) + Ay my(t) sin(2nft) Therefore, > 4, +58 ee 1 % SCH) = 3PM Cet dM jCC40,)] + 35 Mol Pf IM t4f,)) where M,(f) = Flm(t)] and Mo(f) = Flm(t)], The spectrum of the received signal is therefore RCE) H(f)S(f) 1 (2) MCE day (Pef Ie FMS = 5 5 M(tet)I To recover m,(t), we multiply r(t), the inverse Fourier transform of R(f), by cos(2mft) and then pass the resulting output through a low-pass filter, producing a signal with fhe following spectrum FUr(tycos(2aft)] = F (R(faf,)4R(f40,)] A ce NCEE OM (20) + MCE) + FMolter,) - $Me) A + TE HCPeE DMC) + My (te2e.) + F Ma(t) 5 Mylt42r,)1 a The condition H(f,+f) = H#(f,-f) is equivalent to H(f+f,)=H(f-£,)s this follows from the fact that for a real-valued impulse response h(t), we have H(-f)-H#(f), Hence, substituting this condition in Eq. (1), we get A c Fr(t)eos(2nf,t)] = 5° H(f-£,)M,(£) ry ce 1 1 + Pe HLM (P20) + F Np (fF, )4M (F420, - 5 Mp(f42f,)1 The low-pass filter output, therefore, has a spectrum equal to (A,/2) H(f-f,)M,(f). Similarly, to recover m(t), we multiply r(t) by sin(2mf.t), and then pass the resulting signal through a low-pass filter. In this case, we get an output with a spectrum equal to (A,/2) H(f-£,)M)(f). a66 Problem 2.13, When the local carriers have a phase error 4, ve may write cos(2af,t + 4) = cos(2nf.ticos ¢- sin(2xf,t) sin ¢ In this case, we find that by multiplying the received signal r(t) by cos(2af.t+4)s and passing the resulting output through a low-pass filter, the corresponding low-pass filter output in the receiver has a spectrim equal to (A./2) H(f-£,) [cos M,(f) - sing NQ(f)]. This indicates that there is cross-talk at the demodulator outputs. 66Problem 2.14 ‘The transmitted signal is given by 3() = Apm,(t)cos (2 ft) + Apma(1)sin(2n ft) = A(LVo + mj(t) + m,(t)]cos(2nf,t) + A[m,(t) ~m, (1) ]sin (2m f,2) (a) The envelope detection of s(0) yields Yilt) = Agal(Vo +m (4) +m,(1))° + (omit) = m,(1)° mt) — m,(t) \2 = A(Vg + m,(t)+m,(0) 1 aa) To minimize the distortion in the envelope detector output due to the quadrature component, we choose the DC offset Vo to be large. We may then approximate yy(1) as y(t) = A(Vo + m1) +m,(0)) which, except for the DC component A, Vo, is proportional to the sum m(t) + m,(t). (b) For coherent detection at the receiver, we need a replica of the carrier A,cos(2n/,t). This requirement can be satisfied by passing the received signal s(¢) through a narrow-band filter of mid-band frequency f.. However, to extract the difference m,(t) - m,(2), we need sin(2rf,f), which is obtained by passing the narrow-band filter output through a 90°-phase shifter. Then, multiplying (0) by sin(2ryt) andt low-pass filtering, we obtain a signal proportional to mt) - m,(0) (©) To recover the original loudspeaker signals m1) and m,(t), we proceed as follows: + Equalize the outputs of the envelope detector and coherent detector. + Pass the equalized outputs through an audio demixer to produce mt) and m,(0), o7Problem 2.15 (a) s(t) = A.C + kym(1))cos (2m ft) (1, te id 5 |cos(2m ft) \ ler) To ensure 50 percent modulation, ka = I, in which case we get s(t) = alt+s 1 )eos(2nf.) +P (b) s(t) = Agn(t)cos(2nf,.t) A, S cos(2n ft) Ler Ac, (c) s(t) = q Lam(z)cos (2m ft) — Hi(r)sin(27f.t)) 7 cos(2mf,.t)- 1 sin(2nf,1) Sram aerate sy = Sf 1 cos(2me ft) + sin y_0)] thier? ier As an aid to the sketching of the modulated signals in (c) and (d), the envelope of either SSB wave is _if?ser 1 fa 00 aaa Wie? Plots of the modulated signals in (a) to (d) are presented in Fig. 1 on the next page. 68= 10a) 6 4 ~2 0 2 4 6 8 10 69Problem 2.16 Consider first the modulated signal s(t) = fim(t)cos(2m ft) ~ sin(t) sin (2m f,.1) a Let S(f) = F[s()], Mi) = Flm(2)], and M(f) = f[sa(t)] where r(r) is the Hilbert transform of the message signal m(j). Then applying the Fourier transform to Eq. (1), we obtain SU) = MEF) # MU + fo)I= LMS F-) MUS + £2] @ From the definition of the Hilbert transform, we have Mf) = ~jsgn(f)M(f) where sgn({) is the signum function. Equivalently, we may write HF f= sen f-FMU-F.) HHS) = sen + SIMS +S (i) From the definition of the signum function, we note the following for f> 0 = and f> fi: sen(f- fo) sen(f+f,) = +1 Correspondingly, Eq. (2) reduces to SCA) = HMC.) 4 MCF ef 1+ IMC -F)-MF +f] 1 = 5MUE-f) In words, we may thus state that, except for a scaling factor, the spectrum of the modulated signal (¢) defined in Eq. (1) is the same as that of the DSB-SC modulated signal for f > f,. Gi) For f> 0 and f< f,, we have 70sen(f-f,) = -l sen(f+f,) = +1 Correspondingly, Eq. (2) reduces to S(f) GIMS-L¢ MF + f+ LMU fe) - MEF] =0 In words, we may now state that for / 0. This result also holds for f< 0, the proof for which is left as an exercise for the reader. Following a procedure similar to that described above, we may show that the modulated signal s(t) = (eos (2n ft) + (0) sin(2nf.1) @) represents a single sideband modulated signal containing only the lower sideband. nNProblem 2.17 ‘An error Af in the frequency of the local oscillator in the demodulation of an SSB signal, measured with respect to the carrier frequency f,, gives rise to distortion in the demodulated signal. Let the local oscillator output be denoted by A, cos(2n(f, + Aft). The resulting demodulated signal is given by (for the case when the upper sideband only is transmitted) volt) = ; A,A, [m(t)cos(2nAft) + m(t)sin(2nAft)] ‘This demodulated signal represents an SSB wave corresponding to a carrier frequency Af. ‘The effect of frequency error Af in the local oscillator may be interpreted as follows: @ (b) If the SSB wave a(t) contains the upper sideband and the frequency error Af is positive, or equivalently if s(t) contains the lower sideband and Af is negative, then the frequency components of the demodulated signal v,(t) are shifted inward by the amount Af compared with the baseband signal m(t), as illustrated in Fig. 16). If the incoming SSB wave s(t) contains the lower sideband and the frequency error Af is Positive, or equivalently if s(t) contains the upper sideband and Af is negative, then the frequency components of the demodulated signal v,(t) are shifted outward by the amount Af, compared with the baseband signal m(t). This is illustrated in Fig. 1¢ for the case of a baseband signal (e.g., voice signal) with an energy gap occupying the interval -f, _# The output of the lower first product modulator has the spectrum: J Gat E-§) EMEP A i The output of the upper low pass filter has the spectrum: aM, G+f) 14The output of the lower low pass filter has the spectrum: ~ => M_(f. ay MF) Sth f OL ~4.£ mo “ ! 0 “3 M$) Tne output of the upper second product modulator has the spectrum: iM, Ge fed) 4 1 io, (44-2) fl . 4 Fog tM G-£-4) ME Sad) ‘ @ et i aa oo <_—~I___+ oF The output of the lower second product modulator. has the spectrum: 2 4f-£) Mg bef) 4 Ma fh au, §-4) Adding the two second product modulator outputs, their upper sidebands add constructively while their lower sidebands cancel each other. (c) To modify the modulator to transmit only the lower sideband, a single sign change is required in one of the channels. For example, the lower first product modulator could multiply the message signal by -sin(2nf.t). Then, the upper sideband would be cancelled and the lower one transmitted. 15Problem 2.19 m(t) Product El sa High-pass 2} Product: 3 (or /OW~Pass i) modulator filter [--=|modulator filter cos (2nf t) cos [2m (f+, )t] (a) The first product modulator output is v(t) = m(t) cos(2xf,t) The second product modulator output is v3(t) = volt) cosl2mt +f.) t] The amplitude spectra of mt), v4(t), vp(t), v3(t) and s(t) are illustrated on the next page: . We may express the voice signal mt) as mt) = Fim (t) + m¢t)1 where m,(t) is the pre-envelope of m(t), and m_(t) * m,(t) is its complex conjugate. The Fourier transforms of m,(t) and m(t) are defined by (see Appendix 2) ae), £>0 M(t) = oO, £0 Mf) = a ar), feo Comparing the spectrum of s(t) with that of m(t), we see that s(t) may be expressed in terms of m,(t) and m(t) as follows: 1 1 s(t) = gm (toexp(-Jenf,t)+ g m (texpl jex,t) Bo f= of Lint )+ 580 t)Jexp(-Jenf t+ Bim t)-5ACt dLexpl Jen) n(teos(2aft+ yf A(t) sin(2af,t) (>) With s(t) as input, the first product modulator output is v(t) = s(t) cos(2af,t) 16Ince) | Ince | 1 81 C4Problem 2.20 (a) Consider the system described in Fig. 1a, where u(t) denotes the product modulator output, as shown by u(t) = Agm(t)cos(2n ft) ane product | (| Bandpass Modulated a modulator titer signa! ae HO 1) A005 (28) ) Modulates psu signal Product |_| Lowpass Demeaistes 0) modulator ‘iter “A Aye0s (nf) o Figure 1: (a) Filtering scheme for processing sidebands. (b) Coherent detector for recovering the message signal. Let H(f) denote the transfer function of the filter following the product modulator. The spectrum of the modulated signal s(7) produced by passing u(t) through the filter is given by S(f) = U(NH(f) A = ZIMF-f+MUF + SHY) wm where M(f) is the Fourier transform of the message signal m(?). The problem we wish to address is to determine the particular H(f) required to produce a modulated signal s(¢) with desired spectral characteristics, such that the original message signal m(t) may be recovered from s(t) by coherent detection, The first step in the coherent detection process involves multiplying the modulated signal s(t) by a locally generated sinusoidal wave A’,cos(2m/,t), which is synchronous with the carrier wave A.cos(2n/,f), in both frequency and phase as in Fig. 1b. We may thus write 78vn) cos (2mf,.t)s(t) ‘Transforming this relation into the frequency domain gives the Fourier transform of v(1) as av VP) = SASF. + SF + F1 @ Therefore, substitution of Eq. (1) in (2) yields AA’, VP) = SMA -F.) +H + Fd IM(f-2F )H(f -f.)+M(f + 2f HS + £21 @) (b) The high-frequency components of v(t) represented by the second term in Eq, (3) are removed by the low-pass filter in Fig. 1b to produce an output v,(z), the spectrum of which is given by the remaining components: AA Vif) = AM NLAS -F.) + HF +f) @) For a distortionless reproduction of the original baseband signal m(#) at the coherent detector output, we require V,(/) to be a scaled version of M(/). This means, therefore, that the transfer function H(f) must satisfy the condition AS-f)+H(F +f.) = 2H(f) (6) where H(,), the value of H(f) at f= /,, is a constant. When the message (baseband) spectrum M(f) is zero outside the frequency range -W < f'< W, we need only satisfy Eq. (5) for values of f in this interval. Also, to simplify the exposition, we set H(f,) = 1/2. We thus require that H1(f) satisfies the condition: Hf-f+H(f +f) -wsf> 1, we may express S(f) as follows : 3 O(tof,) + F sinett(t-r,-0)1 = Esinettft.y), > 0 sf) = A 2 O(f0,) + F sineltitetyeae)) = Z sinelt(ff,)1, Problem 2.27 For SSB modulation, the modulated wave is A s(t) = [m(t) cos(2nt.t) + f(t) sin(2nrt)], the minus sign applying when transmitting the upper sideband and the plus sign applying when transmitting the lower one. Regardless of the sign, the envelope is A a(t) = 527 me(ty + w2cey (a) For upper sideband transmission, the angle, O(t) = anfyt + tan” The instantaneous frequency is, 1 ft) £,(t) = 7 SD sf e MUD ANE) — ACE) atte) © ar(m@(t) + Ce) where ' denotes time derivative. (b) For lower sideband transmission, we have fit). mt) * d Q(t) = ant + tan’ and (ty et.» BUED at{e) = ate) f(t) 2x (m@(t) + H(t) 88Problem 2.28 (a) The envelope of the FM wave s(t) is Syn? sin® 1p ¥ te sin“(ant,t) The maximum value of the envelope is a(t) and its minimum value is nin ‘Therefore, a st = / 148 ‘min This ratio is shown plotted below for 0 < 8 < 0,3: Qmae nin (>) Expressing s(t) in terms of its frequency components: s(t) = A, cos(2nf.t) +36 A, coslan(fi+f dt] - 38 A, coslen(f-f,)t] 89The mean power of s(t) is therefore 2 42,242 ag aS | BAS 2*o +3 2 AS cs ‘c ea+hy The mean power of the unmodulated carrier is which is shown plotted below for 0 <8 < 0.3: 90(e) The angle 6,(t), expressed in terms of the in-phase component, s,(t), and the quadrature component, s(t), ist 05 (t) = anft ji = ft + (esin(2rf,t)1 Since tan""(x) = x= x9/3 4.0.5 ag(t) = anf + pain(ant.t) — & stn3anet) 00) = Sri ot + Bsintentt) ~ 3 7 The harmonic distortion is the power ratio of the third and first harmonics: For 8 = 0.3, Dy = 0.09% n Problem 2.29 (a) The phase-modulated wave is s(t) = A, coslarf.t + kA, cos(2nfgt)] A, coslanf.t + 8, cos(2nf,t)] ¢ COS(2nf gt) coslB, cos(2af,t)] - A, sin(2nfgt) sins, cos(2nt,t)) SD)If 8, $0.5, then costa, cos(anf,t)] = 1 Sin{B, cos(2nf,t)] = 8, cos(2nt,t) Hence, we may rewrite Eq. (1) as s(t) = Ay cos(2nf .t) - 8 Ay sin(2nf .t) cos(2nft) 1 = AS cos(2nft) “2 8D Ay sinl2n(f,+f,)t] 1 ~ 38, A, sinter (t-f,)t] (2) The spectrum of s(t) is therefore ar S(f) = FALLS (ff,) + 6 (#4801 ~ Ty Bp AglOUtty-t,) - Siar sf,)] | 1 ~ Tp Bp Mel (Eat yef,) = S(tef -£,)1 (b) The phasor diagram for s(t) is deduced from Eq. (2) to be as follows: Lower side-frequency carrier anf t m <—— Upper side-frequency The corresponding phasor diagram for the narrow-band FM wave is as follows: 92carrier a, Lower side~frequency Comparing these two phasor diagrams, we see that, except.for a phase difference, the narrow-band PM and FM waves are of exactly the sane form. Problem 2.30 The phase-modulated wave is s(t) = AY coslanf t + a cos(2nf t)) The complex envelope of s(t) is Bt) = Ay expl 8, cos(2nf,t)] Expressing S(t) in the form of a complex Fourier series, we have Blt) =e, exp( j2mnt,t) news where Vey. che ty! 3(t) exp(-j2mnf,t) at nM aye, . vet, FAL ag APL Sp COBC2E GE) — J2mf,t] dt a” ” Let aft = 1/2 -¢. Then, we may rewrite Eq. (1) as A =f exp- A -1/2 £ expt J, sin(g) + ing] do a /2 93The integrand is periodic with respect to ¢ with a period of 2. Hence, we may rewrite this expression as 2 exp(- IE) 5 expt js, sin(g) + jmp] dp Boe p However, from the definition of the Bessel function of the first kind of order n, we have « jo ek £ expld x sing - nie) a6 Therefore, ine ig exp(- AB) JB) We may thus express the PM wave s(t) as s(t) = Re[S(t) expt janf.t)] A, Rel T° J_n(8p) exp(- 9B expt jeentyt) exp( j2nfgt)) ns» lan (Bp) cosl2n(foenf Dt = 5E The band-pass filter only passes the carrier, the first upper side-frequency, and the first lower side-frequency, so that the resulting output is Bolt) = A, Jo(B,) cos(2nt.t) +A, J_1(8,) coslan(f +f, It = 2 1 +A, Jy(B) coslen(t Fyre + 5 = AQ IolB,) cos(2nf gt) +A, J_{(8,) sink2n (tf, )t] A, J4(8,) sinlan(f ft] But 146,) = J4@,) Therefore, y(t) eA JB) cos(2nf t) -Ay J4@p) {sinlan(f +f, )t) + sinl2n(f,-f,)t]) eA, Jo(By) cos(2nft) = 2 A, 946) cos(nf t) sin(arf.t) The envelope of s,(t) equals o4a(t) / 2, +4 (8) 408.) cos“(2nf t) ec” Yo! The phase of s(t) is 2 34(8,) Jol) 1 ot) = tan"! £ cos(2xft)) The instantaneous frequency of s,(t) is ef. + Le dott) nen ete 4 a 2 J (Bp) 5,6,) sin(2nf t) SIC) + 4958, cos“C2nr gt) Problem 2.31 (a) From Table Au.1, we find (by interpolation) that Jp(B) is zero for B= 2.44, 8 = 5.52, B = 8.65, B = 11,8, and so on. (b) The modulation index is oat, etn tn Therefore, Bf, n keg n Since Jg(8) = 0 for the first time when B = 2.44, we deduce that 2.uu x 103 2 = 1.22 x 103 hertz/volt Next, we note that Jo(B8) = 0 for the second time when 8 = 5.52. Hence, the corresponding value of A, for which the carrier component is reduced to zero is 951.22 x 103 5.52 x 103 = 4,52 volts Problem 2.32 For 8 Ig) 340) JQ) Therefore, s9ct) = 1, we have = 0.765 = 0.4m = 0.115 the band-pass filter output is (assuming a carrier amplitude of 1 volt) = 0.765 cos(2nf,t) + 0.44 {coslan(f.+f,)t] - cosl2n(f,-f,)t]) + 0,115 {coslén(f,+2f,)t] + coslan(f,-2f,)t]) » and the amplitude spectrum (for positive frequencies) is [So a 96Problem 2.33 (2) The frequency deviation is Af = kp Ay = 25 x 103 x 2025 x 10° Hz ‘The corresponding value of the modulation index is af | 5 x 10% 10 B= 5 ‘The transmission bandwidth of the FM wave, using Carson's rule, is therefore By = 2f,(148) = 2x100 (145) = 1200 kHz = 1,2 MHz. (>) Using the universal curve of Fig. 336 we find that for a ra Therefore, By = 3x500 = 1500 kHz = 1.5 MHz (c) If the amplitude of the modulating wave is doubled, we find that Af = 1 Miz and p = 10 Thus, using Carson's rule we obtain By = 2x100 (1410) = 2200 xHz = 2,2 MHz Using the universal curve of Fig. 3:36, we get By apt 25 and By = 2.75 Miz. (4) If £, is doubled, g 2.5. Then, using Carson's rule, By = 1.4 Miz. universal curve, By/af and By = 4af = 2 Miz. 7 Using theProblem 2.34 (a) The angle of the PM wave is o(t) Aafgt + ky mt) arf gt + ky A,cos(2nf,t) ant.t + 8, cos(2rf,t) where B, =k, Aye The instantaneous frequency of the PM wave is therefore (0) ale ar at f. - 8, fy sin(2nf,t) We see that the maximum frequency deviation in a PM wave varies linearly with the modulation frequency f,. Using Carson's rule, we find that the transmission bandwidth of the PM wave is approximately (for the case when 6, >> 1) By = 2 + 8p fq) = 2y(1 +8,) = 2, 8 Tais shows that By varies linearly with f,. (>) In an FM wave, the transmission bandwidth By is approximately equal to 2af, if the modulation index 8 >> 1. Therefore, for an FH wave, By is effectively independent of the modulation frequency f,. Problem 2.35 ‘The filter input is v(t) = g(t) s(t) g(t) cos(2nt,t = set?) The complex envelope of v,(t) “is F(t) = g(t) explagekt®) ‘The impulse response h(t) of the filter is defined in terms of the complex impulse response h(t) as follows h(t) = Reh(t) exp( jarft)] with h(t) = costanf st + wkt™), we have Ree) = expt jnit) aof the The complex envelope¥filter output is therefore (see Appendix 2) Volt) =gRceoee V(t) a(t) exp(-jnkr®) expl jak(ter)]°dr via f 1 2a B exp jake) f g(t) exp(-Jenktr) dr 1 2. B exp(inkt) G¢Kt) Hence, Wigton =F tect This shows that the envelope of the filter output is, except for the scale factor of 1/2, equal to the magnitude of the Fourier transform of the input signal g(t), with kt playing the role of frequency f. Problem 2.36 The overall frequency multiplication ratio is n= 2x3 = 6 Assume that the instantaneous frequency of the FM wave at the input of the first frequency multiplier is fyy(t) = fy + af cos(2sf gt) The instantaneous frequency of the resulting FM wave at the output of the second frequency multiplier is therefore fy p(t) = nf, + ndf cos(2nft) Thus, the frequency deviation of this FM wave is equal to nAf = 6x10 = 60 kHz and its modulation index is equal to nat _ 60 nat . 0 2 a2 t, 3 The frequency separation of the aijacent side-frequencies of this FM wave is unchanged at f, = 5 kHz. ™ 99Problem 2.37, (a) Figure 1 shows the simplified block diagram of a typical FM transmitter (based on the indirect method) used to transmit audio signals containing frequencies in the range 100 Hz to 15 KHz. The narrow-band phase modulator is supplied with a carrier signal of frequency f; = 0.2 MHz by a crystal-controlled oscillator. The desired FM signal at the transmitter output is to have a carrier frequency f= 100 MHz and a minimum frequency deviation Af'= 75 kHz. In order to limit the harmonic distortion produced by the narrow-band phase modulator, we restrict the modulation index Bj of this modulator to a maximum value of 0.3 radians. Consider then the value i; = 0.2 radians, which certainly satisfies this requirement. The lowest modulation frequencies of 100 Hz produce a frequency deviation of Af, = 20 Hz at the narrow-band phase modulator output, whereas the highest modulation frequencies of 15 KHz produce a frequency deviation of Af, = 3 kHz. The lowest modulation frequencies are therefore of immediate concern, as they produce a much lower frequency deviation than the highest modulation frequencies. The requirement is therefore to ensure that the frequency deviation produced by the lowest modulation frequencies of 100 Hz is raised to 75 KHz. aera ro aowbona] | Freaerey Fraqueney | soa Hd ingatoe Lol ne Lol fattet Lol ier Loy moter bo meta [| at ey cyst cry Figure 1 To produce a frequency deviation of Af = 75 kHz at the FM transmitter output, the use of frequency multiplication is obviously required. Specifically, with Af, = 20 Hz and Af = 75 kHz, ‘we require a total frequency multiplication ratio of 3750, However, using a straight frequency multiplication equal to this value would produce a much higher carrier frequency at the transmitter output than the desired value of 100 MHz. To generate an FM signal having both the desired frequency deviation and carrier frequency, we therefore need to use a two-stage frequency ‘multiplier with an intermediate stage of frequency translation as illustrated in Fig. 1. Let m and ny denote the respective frequency multiplication ratios, so that Af _ 73000 af 20 nyny = 3750 a 100‘The carrier frequency nf; at the first frequency multiplier output is translated downward to (J - nyfi) by mixing it with a sinusoidal wave of frequency f, = 95 MHz, which is supplied by a second crystal-controlled oscillator. However, the carrier frequency at the input of the second frequency multiplier is required to equal f./ny. Equating these two frequencies, we thus get Soom fy = E ny Hence, with fy = 0.1 MHz, fy = 9.5 MHz, and f, = 100 MHz, we have 9.5-0.1n, = 1 2) nm Solving Eqs. (1) and (2) for n, and 9, we obtain ny =75 ny =50 (b) Using these frequency multiplication ratios, we get the set of values indicated in the table below: Table -Values of Carrier Frequency and Frequency Deviation at the Various Points in the Wide-band Frequency Modulator of Fig. 1 At the First ‘At the Second At the Phase Frequency Frequency Modulator Multiplier | Atthe Mixer | Multiplier Output Output Output Output Carrier 0.1 MHz 7.5 MHz 2.0 MHz, 100 MHz frequency Frequency 20 He 1.5 kHz 1.5 kz 15 kHz deviation 101Jem 2.3: (a) Let L denote the inductive component, ¢ the capacitive component, and Cy the capacitance of each varactor diode due to the bias voltage V, acting alone. Then, we have 7 -12 Cy = 100 vo'/? pF and the corresponding frequency of oscillation is awl (Cat 72) Therefore, 25/200 x 10 Solving for Vp, we get Vp = 3-52 volts (b) The frequency multiplication ratio is 64. Therefore, the modulation index of the FM Wave at the frequency multiplier input is 0,078 Tris indicates that the FM wave produced by the combination of L, C and the varactor diodes is a narrow-band one, which in turn means that the amplitude A, of the modulating wave is snall compared to V,. We may thus express the instantaneous fFequency of this FM wave as follows: 12 1224 172 ft) = & 200 x 106 (100 x 1071? + 50 x 107"? 13.52 + AL silent ty 7 An _ 10 1/2 ,-1/2 = (1 + 0,266 (1 +3755 sin(ant tyy7/*) even 7 A ~ 10 m See cee arson ame “1/2 = 10° (1 ~ 0,03 A, sin¢amrt) 37/2 = 10° (1 + 0,015 A, sin(ent,t)) 102js “ith @ modulation index of 0.078, the corresponding value of the frequency deviation 8 ate ee, 0.078 x 104 He Therefore, 0.015 A, x 10° = 0,078 x 10" where A, is in volts. Solving for 4,, we get Ay = 52 x 1079 volts. in Problem 2.39 ‘The transfer function of the RC filter is non) = ae If 2xfCR << 1 for all frequencies of interest, then we may approximate H(f) as H(f) = j2nfCR However, multiplication by j2xf in the frequency domain is equivalent to differentiation in the time domain. Therefore, denoting the RC filter output as v,(t), we may write = or S86) v(t) = oR SE A t = OR Ge (A, cosl2net + 2k, a mt) dt}} t WOR AL2nf, + 2tkpm(t)] sinl2nt.t + 2tky is mt) dt] The corresponding envelope detector output is k f volt) = 2xfcr A,|1 + Fae] Since k,{m(t)] < , for all t, then Ke vat) = 2nfcR ALL + pe mt} Which shows that, except for a de bias, the output is proportional to the modulating signal m(t). 103Problem 2.40 The envelope detector input is v(t) = s(t) = s(t-T) FA, cosl2af te $(t)] - A, cosl2nf (t-T) + ¢(t-T)) nf (2-7) + g(t) + Mt) 2af T + o(t) - g(t-T) = -2A, sinl- J sin J a e 2 2 where o(t) = 8 sin(2et,t) The phase difference g(t) - g(t-T) is ot) ~ (tT) = B sin(2af,t) - 8 sinl2nf,(t-T)) @lsin(2nf,t) - sin(2nf,t) cos(2mf,T) + cos(2nf,t) sin(2rf,T)] Blsin(2uet) - sin(2mf,t) + 2nf,T cos(2mf.t)) 2nA€T cos(2nft) where af: af. Therefore, noting that 2uf.T = 1/2, we may write ant T + Ht) - tT) sint- z J = sinlef.T + xAfT cos(2nft)) = sinl] + "eT cos(2ar,t)] 1 ¥2 cost rafT cos(2nft)] + v2 sin[ wAfT cos(2nf,t)] V2 + V2 wAeT cos(2nt,t) where we have made use of the fact that nAfT << 1, We may therefore rewrite Eq. (1) as v(t) = -2/2 ALC1 + wAfT cos(2afgt)) sintaf,(2t-7) + #0) + oCt-T) Accordingly, the envelope detector output is a(t) = 2 72 A [i+nafT cos(2af,t)] Which, except for a bias term, is proportional to the modulating wave. 104Problem 2.41 (a) In the time interval t-(T4/2) to t+(T4/2), assume there are n zero crossings. ‘The phase difference is 0,(t4T,/2) - 0(t-T,/2) = nt t O((t) = 2nf t + 2mky vs mt) dt. Also, the angle of an FM wave is Since m(t) is assumed constant, equal to my, 0j(t) = 2nf jt + 21kpmyt. Therefore, O,(b4T 4/2) ~ OAT 4/2) = (2, + 2mkym,) [t+T,/2 = (t=T,/2)]. = (xf, + 2rk—my) T,. But 48, (t) f(t) 2 Fp = amr, + Ankeny. Thus, 6, (aT 4/2) - Oy (t-T4/2) = f(t) Tye But this phase difference also equals nt. So, f(t) T, = on and f(t) = n/t, (b) For a repetitive ramp @s the modulating wave, we have the following set of waveformsLimiter output Pulse outputProblem 2.42, The complex envelope of the modulated wave s(t) is Blt) = alt) expljg(t)] Since a(t) is slowly varying compared to exp{jp(t)], the complex envelope S(t) is restricted effectively to the frequency band - B,/2 < f < B,/2. An ideal frequency discriminator consists of a differentiator followed by an envelope detector. The output of the differentiator, in response to 8(t), is F(t) = Ge He) = & fale) expljg(t)]) = SBD cor sgcty] + 3 28 acty explig(t)) a(t) expt so(t)) Catgy S202 + 3 $8) Since a(t) is slowly varying compared to #(t), we have 1 da(t) | > tetey SSE agit) I at Accordingly, we may approximate 7,(t) as ~ =G do(t) T(t) = 9 at) SE expr ject] However, by definition t Ht) = 2mke J mt) dt ° Therefore, T(t) = somky a(t) mt) expl jo(t)] Hence, the envelope detector output is proportional to a(t) m(t) as shown by I(t) 1 = 2nke a(t) mt) Problem 2.43 (a) The limiter output is 2(t) = sgnfa(t) cosl2nf.t + 9(t)]) 107Since a(t) is of positive amplitude, we have 2(t) = sgnfeost2nt.t + o(t)1) Let Vit) = 2afgt + g(t) Then, we may write senfeos ¥] = = ¢, exp(jny) ni ® Fy F santeos ¥ expl-sny) dv 1 272 1 FL G1 expleinpay + Ly (41) expl-gny) ay a 7 7 a ane 7 + £1 expl-iny ay ar ye eg = Faagay (exp Bb sexp( nnd sexp (8%) expat) ~exp(-jnndeexp(Hl85} Lire sinc®h sin(nt)] 2 (n-1)/2 | cn ’ n odd 9, n even If nz0, we find from Eq. (1) that e,20. Therefore, senteos yl = 2 2 2 cat)? exocjnyy eo n odd mre 4 (-1) e- £ cost p(2k+1)] F ly Beet We may thus express the limiter output as 108yk ae a(t) = 4 + eo cosl2nf .t(2k+1) + o(t) (2k+1)] @) (b) Consider the term cosl2nr t(2k+1) + O(t) (2k+1)] = Relexpl Jnr t(2k+1)Jexpl 4(t) (2k+1)]} 241) Re(expl J2nt,t(2e+1) Iexptj4(t)) 1 The function exp[{j¢(t)], representing the complex envelope of the FM wave with unit amplitude, is effectively low-pass in nature. Therefore, this term represents a band-pass Signal centered about #f,(2k+1). Furthermore, the Fourier transform of (explo(t)]}**! is equal to that of expljq(t)] convolved with itself 2k times. Therefore, assuming ee explj@(t)] is limited to the interval -B,/2 < f < By/2, we find that (exotsoces} “te limited to the interval (By /2) (+1) aaa (By/2) (241). Assuming that f > Bor as is usually the case, we find that none of the terms corresponding to values of k greater than zero will overlap the spectrum of the term corresponding to ks0, Thus, if the limiter output is applied to a band-pass filter of bandwidth By and midband frequency f,, all terms, except the term corresponding to k=0 in Eq. (2), are removed by the filter. ‘The resulting filter output is therefore vit) =F costantst + t)) We thus see that by using the amplitude limiter followed by a band-pass filter, the effect of amplitude variation, represented by a(t) in the modulated wave s(t), is completely removed. Problem 2.44 (a) Let the FM wave be defined by t s(t) = Ay cosl2nf.t + 2nky a mt) dt] Assuming that f, is large compared to the bandwidth of s(t), we may express the complex envelope of s(t) as 7 t s(t) = AS expl j2mk, 4 mt) dt] But, by definition, the pre-envelope of s(t) is © Rppencliz 2) 109s(t) = S(t) exp(j2nt,t) s(t) + 5 8(t) where 8(t) is the Hilbert transform of s(t). ‘Therefore, t s(t) + J8(t) = Ay explj2mk, J, me) dtd expt sentt t t Aleosl2nf.t + ark, Fyate) dtl + J sinlant,t + 2aky oo Equating real and imaginary parts, we deduce that t sinl2nf t+ 2nke f mt) dt] a) ce a Blt) = (>) For the case of sinusoidal modulation, we have m(t) =A, cos(2af.t) The corresponding FM wave is s(t) cosl2nf.t + 8 sin(2af,t)] where B= Ke A, Expanding s(t) in the form of a Fourier series, we get ae 4,(8) coslan(f,+nf,t] s(t) c Noting that the Hilbert transform of cos{2(f.+nf,,)t] is equal to sint2m(f.snf_)t], and using the linearity property of the Hilbert transform, we find that the Hilbert Eransform of s(t) is TE J,(8) sinlaw(r ent.) t] B(t) = ni =A, sinf2ne t + 6 sin(2nf_t)] This is exactly the same result as that obtained by using Eq. (1). In the case of sinusoidal modulation, therefore, there is no error involved in using Eq. (1) to evaluate ‘the Hilbert transform of the corresponding FM wave. Problem (a) The modulated wave s(t) is s(t) = expl-@(t)] coslanf.t + ¢(t)] 110Refexp(-9t)] expljanf.t + j9(t)]) Refexpl s2nfgt + JC g(t) + 59(t))]) Relexpl j2af.t + J9,(t)1) a where ¢,(t) is the pre-envelope of the phase function g(t), that is, $, CE) = g(t) + 39Ct Expanding the exponential function exp{jo,(t)] in the form of an infinite series: expl jo, (t)) @ Taking the Fourier transform of both sides of this relation, we may write n Flexpljg,(t)]) = 2 J erg%cty) + Bente For n>2, we may express ¢7(t) as the product of ¢,(t) and g@7"(t). Hence, Fatty] = @ COMEFLt)D where 6,(t)= @ (f), and %denotes convolution. Since @(f) = 0 for f < 0, it follows that for all n > 0, FLAC) = 0, for f <0 Hence, Flexplje,(t)]} = 0 © for £< 0 By using the frequency-shifting property of the Fourier transform, it follows that Flexplse,(t)1 exp(J2negty} = 0 for f < f, @) From Eq. (1), © a(t) = d fexplsoetgt + 39,(t1 + expl=g2nt gt = j¢n(t)) where g(t) is the complex conjugate of 4,(t). Therefore, Fls(t)] = 3 Flenpl iar gt + 39,(t))) + 3 Flexpl-s2ngt - Soet) Applying the conjugate-function property of the Fourier transform to £q. (3), we find that Flexpl-jemt.t - j¢ht)]) = 0, for f> ~ ilus Hence, it follows that the spectrum of s(t) is zero for -f, < f < f,. However, this Spectrum is of infinite extent, because the expansion of s(t) contains an infinite number of terms, as in eq. (2). (b) With $(t) = 8 sin(2af,t), we find that (t) = -8 cos(2nf,t) Therefore, 4,(0) = B sin(2at,t) = 38 cos(2at,t) zo JBloos(2nft) +j sin(2nft)] = = J8 expl nf.) Hence » expl J¢,(t)] = explB exp( j2nf,t)] = gn a ee am: . ae ar exP( Semnf, t) ‘The modulated wave s(t) is therefore s(t) = Relexp( j2nf.t) expl Jo, (t)1} 2 an = Relexp( janf.t) oe = oon = Ret 2 BF expljen(tent,)t]} n=0 cosl2n(f nf, )t]Problem 2.46 Alter passing the received signal through a narrow-band filter of bandwidth 8kH1z centered on Sfe= 200kHz, we get x(t) = Agm(t)cos(2mf,t) + n’(t) = Agn(t)cos(2nf 1) +1’ (t)cos(2nf,t)—n' (d)sin(2nf.1) Agn(t) + ,(1))cos(2nf 1) —n’g(0)sin(2nf,2) where n’(1) is the narrow-band noise produced at the filter output, and n’,(r) and n“9(t) are its in-phase and quadrature components. Coherent detection of x(t) yields the output v(t) = Agm(t) +n’ (0) The average power of the modulated wave is, AP 10W where P is the average power of m(2). To calculate the average power of the in-phase noise component n’,(1), we refer to the spectra shown in Fig. 1 Part (a) of Fig. 1 shows the power spectral density of the noise n(), and a superposition of the frequency response of the narrow-band filter. Part (b) shows the power spectral density of the noise n’,(t) produced at the filter output. + Part (c) shows the power spectral density of the in-phase component n’,() of n’(t) Note that since the bandwidth of the filter is small compared to the carrier frequency f,, we have approximated the spectral characteristic of n’(r) to be flat at the level of 0.5 x 10% watts/FHz. Hence, the average power of n’,(1) is (from Fig. Ic): (10° watts/Hz) (8 x 10) = 0.008 watts The output signal-to-noise ratio (SNR) is therefore 10 = 12 o.o08 = 125° Expressing this result in decibels, we have an output SNR of 31 dB. M3Problem 2.47 From Problen 5. 38, we note that the quadrature components of a narrow-band noise have autocorrelations : Ry (4) = Ry (4) = Ryle) cos(2nfit) + Ry(t) sin(2nf.r) T Q where Ry(t) is the autocorrelation of the narrow-band noise, Ry(t) is the Hilbert transform of Ry(t), and f, is the band center. The cross-correlations of the quadrature components are Ry y (= NN, 18 (a) For a DSBSC system, NN = Ry(t) sin(2nfyt) ~ Ry(t) cos(2nf,t) By (0) = Ry (0) = Ryle) cosCent) + Rye) sin(2xegn) Ry y (1) = Ry y Ce) = Ryle) sin(2net) - Ry(t) cos(2nf,t) ra Qt _ Where f, is the carrier frequency, and Ry(t) is the autocorrelation function of the narrow-band noise on the interval f,-W eALIT + kamtI1) < 645 then, with a probability greater than 1-6, we may say that yO) = ETAL + AS ey met) + mct)]2)72 That is, the probability that the quadrature component n,(t) is negligibly small is greater than 1 - 54. (b) Next, we note that if ke m(t) < -1, then we get overmodulation, so that even in the absence of noise, the envelope detector output is badly distorted. Therefore, in order to avoid overmodutation, we assume that k, is adjusted relative to the message signal m(t) such that the probability PAL + AL ky mt) + ny(t) <0) Then, the probability of the event yt) = ASET +k, mt) + nt) for any value of t, is greater than (1 = 6)(1 - 82). (c) When 6, and 6, are both small compared with unity, we find that the probability of the event, y(t) = ALL1 + ky m(t)] + no (t) for any value of t, is very close to unity. Then, the output of the envelope detector is approximately the same as the corresponding output of a coherent detector. Pr 2.52 The received signal is 123x(t) = A, cos(2nf.t) + n(t) A, cos(2mf.t) +n (t) cos(2mf.t) - ng(t) sin(2ef.t) (a, +n (t)] cos(2nf.t) ~ n.(t) sin(2nf.t) ‘The envelope detector output is therefore a(t) 2g 24) 172 {Ag + ney}? n’ct)) For the case when the carrier-to-noise ratio is high, we may approximate this result as a(t) =A, + ny (t) The term A, represents the useful signal component. ‘The output signal power is thus A®. ‘The Power spectral densities of n(t) and n{(t) are as shown below: s,(£) 2 t_ {Xo = o WW The output noise power is iW. The output signal-to-noise ratio is therefore (SNR)g = 124Problem 2.53 (a) From Section 1.12 of the textbook we recall that the envelope r(¢) of the narrow-band noise n(t) is Rayleigh distributed; that is fel) = fon] oy 20% 2 2 where of is the variance of the noise n(f). For an AM system, the variance oy is 2WNo. Therefore, the probability of the event that the envelope R of the narrow-band noise n(f) is large compared to the carrier amplitude A, is defined by P(R2A,) = f Sp(r)ar " 0) Define the carrier to noise ratio as average carriet_power bandwidth of the modul average noise por d message signal Since the bandwidth of the AM signal is 21, the average noise power in this bandwidth is 2WNo. The average power of the carrier is A2/2. The carrier-to-noise ratio is therefore Ar ° = TWN, ® (b) We may now use this definition to rewrite Eq, (1) in the compact form P(R2A,) = exp(-p) 4) Solving P(R > A,) = 0.5 for p, we get 125p= log? = 0.69 Similarly, for P(R > A. 0.01, we get p=logl00=4.6 ‘Thus with a carrier-to-noise ratio 10log 90.69 = -1.6 dB, the envelope detector is expected to be well into the threshold region, whereas with a carrier-to-noise ratio 10logi94.6 = 6.6 dB, the detector is expected to be operating satisfactorily. We ordinarily need a signal-to-noise ratio considerably greater than 6.6 dB for satisfactory intelligibility, and therefore threshold effects are seldom of great importance in AM receivers using envelope detection. Prol 54 (a) Following a procedure similar to that described for the case of an FM system, we find that the input of the phase detector is v(t) =A, costanf.t + o(t)] where nat) at) = ky mt) + c with ng(t) denoting the quadrature noise component. The output of the phase discriminator is therefore, nig(t) yt) = ky mt) + c The message signal component of yt) is equal to k, m(t). Hence, the average output signal power is Ko P, where P is the message signal power. With the post detection low-pass filter following the phase detector restricted to 126the message bandwidth W, we find that the average output noise power is 2HNy/A2, Hence, the output signal-to-noise ratio of the Pi system is x? pa? e AW (bv) The channel signal-to-noise ratio of the PM system is the same as that of the corres- ponding FM system. That is, (SNR)y = a (SNR )g = ago 0 Tne figure of merit of the PM system is therefore equal to ky P. For the case of sinusoidal modulation, we have m(t) = A, cos(2nf,t) Hence , a2 peat z The corresponding value of the figure of merit for a PM system is thus equal to 3 sy Where 8, = ky Aq. On the other hand, the figure of merit for an FM system with sinusoidal 3 2 modulation is equal to $ 6°. We see therefore that for a specified phase deviation, the FM systen is 3 times as good as the PM system. Problem 2.55 (a) The power spectral densities of the original message signal, and the signal and noise components at the frequency discriminator output (for positive frequencies) are illustrated below: Spectral densi of message Signa £ (kHz) Spectral density | ot signal component ot discriminator out ek £ (kHz) 4 8 32 ie 20 24 28 32 36 40 44 48 127Spectr density eb neve component of diserim: nator | i i 1 ube 1 ! 1 £ (kHz) ° 48 (b) Each SSB modulated wave contains only the lower sideband. Let A, and kf) denote the amplitude and frequency of the carrier used to generate the kth modulated wave, where — 4 kHz, and k = 1, 2, ..., 12. Then, we find that the kth modulated wave occupies the frequency interval (k - 1)f < If] < kfg. We may define this modulated wave by A My sy(t) = 5 m(t) cos(2nkf gt) + 5 A(t) sim(2nkf gt) where m(t) is the original message signal, and fi(t) is its Hilbert transform. Therefore, the average power of 5,(t) is A? P/u, where P is the mean power of mt). We may express the output signal-to-noise ratio ; for the kth SSB modulated wave as follows: ee BAL keCAD P/4) (SNR), of 3p fl 2N(ktS = (k - 19°85] 2 4242 Bae aR KEP 303x2 BN of 5 (3k - 3k + 1) where A, is the carrier amplitude of the FM wave. For equal signal-to-noise ratios, we must therefore choose the A, so as to satisfy the condition 2 K Re - eaT a constant for k = 1, 2 see, 12.Problem 2.56 The envelope r(t) and phase y(t) of the narrow-band noise n(t) are defined by x0 = nro + nd eee [ess] aye) For a positive-going click to occur, we therefore require the following: ny(t) = - Ae ng(t) has a small positive value 4 tant 2 |g at a Correspondingly, for a negative-going click to occur, we require nyt) ~ Ag ng(t) has a small negative value 8 tant [MA] 9 ad Ay)Problém 2.57 c V4) ex Mul! Let H(f) be Vi (fVy,(f, or the transfer function of the filter. At low frequencies, the capacitor behaves as an open circuit, Then, R_ LR HOD) # gt Thus, the low frequencies of the input are frequency-modulated. At high frequencies, the capacitor behaves as a short circuit in relation to the resistor. Then, H(t) = s ance, B+ Soare and Vour(t) = RC gS vy Ct) Frequency modulating the derivative of a waveform is equivalent to phase modulating the waveform. ‘Thus, the high frequencies of the input are phase modulated. Problem 2.58 (a) For the average power of the emphasized signal to be the same as the average power of the original message signal, we must choose the transfer function H,,(f) of the pre~ emphasis filter so as to satisfy the relation - 2 LS s Fel? Sy(nar With r s, — tcc 1+ (£/t9)2 o, elsewhere. 5 + £ Hoo(f) = kG + 2) we have w r af Ws (ify 130Solving for k, we get a (b) The improvement in output signal-to-noise ratio obtained by using pre-emphasis in the transmitter and de-enphasis in the receiver is defined by the ratio aw W 2 2 3s f° tH (At? at wee aw? fiat eK? 1 + (4/25) = —W>2 3. 2 3 k°(W/ fy) 7 £ (2) 3U(W/ty) ~ tan™"(W/E9)1 Substituting Eq. (1) in (2), we get 2 at (W/f))° tan” (W/fp) eee eee (3) BECW/fy) ~ tan™'(W/E5)I This result applies to the case when the rms bandwidth of the FM system is maintained the same with or without pre-emphasis. When, however, there is no such constraint, we find from Example 4 of Chapter 6 that the corresponding value of Dis 3 De (Wty) 3L(w/fy) = tan™"(W/fy)I a) In the diagram below, we have plotted the improvement D (expressed in decibels) versus the ratio Wf, for the two cases; when there is a transmission bandwidth constraint and when there is no such constraint: 131Ots> a decibels Problem 2,59 W/hy 1 3.16 10 31.6 100) In a PM system, the power spectral density of the noise at the phase discriminator output (in the absence of pre-emphasis and de-emphasis) is approximately constant. Therefore, the improvement in output signal-to-noise ratio obtained by using pre-emphasis in the transmitter and de-enphasis in the receiver of a PM systen is given by W fat 0 W 2 a Tg) Mae With the transfer function Hy9(f) of the de-enphasis filter defined by Hge(f) = 4 de’ T+ Gt) ' We find that the corresponding value of Dis w i a 0 1+ (£/f9) D= Wy ant tan™'(W/f,) For the case when W = 15 kz, f) = 2.1 kHz, we find that D = 5, or 7 4B, The corresponding value of the improvement ratio D for an FM system is equal to 13 dB (see Example 4 of Chapter 5), Therefore, the improvement obtained by using pre-emphasis and de-emphasis in a PM systen is smaller by an anount equal to 6 dB. 132Problem 2.60 Matlab codes 4 amplitude demodulation Yproblem 2.60, CS: Haykin % Mathini Sellathurai % message signal +t=[0:260] +10-8; ncein(2epiein, +: plot(t, m) xlabel('time (s)") ylabel (Amplitude? ) pause % amplitude modulated signal UFAM_wod(mue,m,ts,f¢); plot(t,u) xlabel (time (s)’) ylabel( Amplitude’) Pause 4% demodulated signal Let, demt]=AM_demod(me,u,ts,te) ; plot(tists, dent) xlabel(’tine (s)’) ylabel(’Amplitude’) axis([0 2.5e-3 0 21) 133function u=AM_mod(mue,m,ts,f¢) % amplitude modulation Yused in problem 2.60, CS: Haykin 4% Mathini Seliathurai % t=[0:1ength(m)-1] ts; cxcos(24pivtc.+t); m_n=n/nax(abs(m)); u=(14muetmn).#¢; 134function [t, env]=AM_denod(mue,m,ts,£c) % Amplitude demodulation Yused in problem 2.60, CS: Haykin % Mathini Sellathurai % ts=1/ts; fsofc=round(ts/tc); n2=Length(m); -zeros(1,round(n2/fsote)); % initializing the envelope (000; % load R €=0.010-6; % capacitor Yadenodulate the envelope 1=0; v(1)=m(1); for sofc:n2-fsofe 1142; v(l)=n(k)sexp(-ts/(R_L#C)/tsofe); % discharging v(LH)=m(keteote); “charging end 4% envelope t =0:fs0fc/2: (Length(v)-1)*£s0f¢/2; 135Answer to Problem 2.60 \A tine () Figure 1: Message signal 136ne ime =) Figure). Amplitude modulated signal me 6) Figureg: Demodulated signal 137Problem 2.61 Matlab codes 4% Problem 2.61 CS: Haykin 4% phase lock loop and cycle slipping %M. Sellathuras Y time interval t0=0;tf=28; % frequency step =0.125 Hz delt=0.125; u0=[0 ~delz+2¥pi]; [e,u)=ode23('1in’ [to tf] ,u0); plot (t,u(:,2)/2/pitdelf); xlabel('Tine (8)") ylabel(t_4 (t), (He)’) pause 4% frequency step =0.51 Hz delf=0.5; u0=[0 -deltepiv2]; Ce ul=ode2a(*1in’, [to t#],uo); plob(t,u(:,2)/2/pitdels); xlabel (Time (s)’); ylabel (fi (t), (Bz)"); pause; 4% frequency step =7/12 liz el#=7/12; 0 ~deltepiv2} "5 Ce ul=ode23(°2in’ , [eo e£],u0); plot (t,u(:,2)/2/pitdelt); xlabel(’Time (s)’}; ylabel(?£i (t), (Hz); pause; % frequency step =2/3 Hz delf=2/3; 0 ~delfepie2)'; [e,ulsode23(?1in’, [to t#],u0); plot (e,u(:,2)/2/pitdels); xlabel(’Time (s)'); ylabel(’fi (t), (Rz)'); 138function uprim =1in(t,u) % used in Problem 2,61, CS: Haykin PLL Transfer function (1+as)/(1+bs), gain K=50/2/pi, natural frequency 1/2/pi damping 0.707 Mathini Sellathurai uprim(2)=-(1/60+1 .2883¢cos(u(1)))eu(2)-sin(a(t)); uprim=uprim’ ;Answer to Problem 2.61 ) Figure (: Variation in the instantaneous frequency of the PLL’s voltage con- trolled oscillator for varying frequency step A f. (a) A f= 0.125 He 1401 “Tone te) Pigure 22 (b) A f= 0.5 Hz mee) Figures: (b) Af = 7/12 He 1412 Figure 4: (b) Af = 2/3 Hz 142CHAPTER 3 Pulse Modulation Problem 3.1 Let 2W denote the bandwidth of a narrowband signal with carrier frequency f,. The in-phase and quadrature components of this signal are both low-pass signals with a common bandwidth of W. According to the sampling theorem, there is no information loss if the in-phase and quadrature ‘components are sampled at a rate higher than 2W. For the problem at hand, we have fe = 100 kHz 2W= 10 kHz Hence, W = 5 kHz, and the minimum rate at which it is permissible to sample the in-phase and quadrature components is 10 kHz. From the sampling theorem, we also know that a physical waveform can be represented over the interval -0 << 00 by 8 = YD 4,0, (0) co) where {@,(f)} is a set of orthogonal functions defined as sin{nf,(¢-n/f,)} 8) = FWA where 1 is an integer and f, is the sampling frequency. If g(#) is a low-pass signal band-limited to W Hz, and f, > 2W, then the coefficient a, can be shown to equal g(n/f,). That is, for f, > 2W, the orthogonal coefficients are simply the values of the waveform that are obtained when the waveform is sampled every L/f, second, As already mentioned, the narrowband signal is two-dimensional, consisting of in-phase and quadrature components. In light of Eq. (1), we may represent them as follows, respectively: gilt) = DY an/F.)0,(2) 143solt) = DY soln/F)0,(0) Hence, given the in-phase samples af +) and quadrature samples sf?) . We may reconstruct oi fe the narrowband signal g() as follows: 8(1) = g(t)cos(27f,t) — go(t)sin(2nf,0) => [s{F) cos(2nf,t) - go sin(2nf0)]0,(8) where f, = 100 kHz and f, > 10 kHz, and where the same set of orthonormal basis functions is used for reconstructing both the in-phase and quadrature components. 144Problem 3.2 (a) Consider a periodic train o(f) of rectangular pulses, each of duration T. The Fourier series expansion of et) (assuming that a pulse of the train is centered on the origin) is given by ct) = YO f, sindinf, TexpGzmnf,t) where f, is the repetition frequency, and the amplitude of a rectangular pulse is assumed to be 1/T (ie., each pulse has unit area). The assumption that f,T>>1 means that the spectral lines (ie., harmonics) of the periodic pulse train e(t) are well separated from each other. Multiplying a message signal g(t) by c(t) yields s(t) = o(t)g(t) = Ef, sine(nf,T) g(t) expG2enf,0) os ‘Taking the Fourier transform of both sides of Eq.. (1) and using the frequency-shifting property of the Fourier transform: EG | oy eee ra @ nese where Gif) = Flg(t)]. Thus, the spectrum S(f) consists of frequency-shifted replicas of the original spectrum G(f), with the nthreplica being scaled in amplitude by the factor f,sine(nf,T). 145() In accordance with the sampling theorem, let it be assumed that « The signal g(t) is band-limited with Gf) =0 for -W2W Then, the different frequency-shifted replicas of G(f) involved in the construction of S(f) will not overlap. Under the conditions described herein, the original spectrum G(f), and therefore the signal g(t), can be recovered exactly (except for a trivial amplitude scaling) by passing s(t) through a low- pass filter of bandwidth W. Problem 3.3, (a) g(t) = sine(200¢) This sinc pulse corresponds to a bandwidth W = 100 liz. Hence, the Nyquist rate is 200 liz, and the Nyquist interval is 1/200 seconds. (b) g(e) = sine” (200t? This signal may be viewed as the product of the sinc pulse sinc(200t}’ with itself. Since multiplication in the time domain corresponds to convolution in the frequency domain, ve find that the signal g(t) has a bandwidth equal to twice that of the sinc pulse sin(200t}; that is, 200 Hz. The Nyquist rate of g(t) is therefore 400 Ha, and the Nyquist interval is 1/400 seconds. (e) g(t) = sine(200e), + sinc”(200e) The bandwidth of g(t) is determined by the highest frequency component of sinc(200t) or sinc2(200t}, whichever one is the largest. With the bandwidth (ive., highest frequency component of) the sinc pulse sinc(200 t) equal to 100 Hz and that of the squared sinc pulse sinc2(200t}) equal to 200 Hz, it follows that the bandwidth of g(€) is 200 Hz. Correspondingly, the Nyquist rate of g(t) is 400 Ilz, and its Nyquist interval is 1/400 seconds. 146Problem 3.4 (a) The PAM wave is s(t) = E (1 + um! (nT) Je(tent,), where g(t) is the pulse shape, and mt(t) = m(t)/A, = cos(2mf,t). The PAM wave is equivalent to the convolution of the instantaneously sampled [1 + um'(t)] and the pulse shape g(t): s(t) = { E [taum'(nt)] Stent.) } xe at) nese = {Ct4nm'(t)1 8 (tnt) } Ye @(t) Tne spectrum of the PAM wave is, SCA) = {CCC some) ye or ocr = By} acer 5 1 2 2 ett) E c6cr = By sume By] 1; m-° Ts Ts For a rectangular pulse g(t) of duration T:0.45s, and with AT = we have: G(f) = AT sine(fT) = sinc(0.45f) For m'(t) = cos(2™fmt), and with ty = 0.25 Hz, we have: mice) = 4 6¢#-0.25) + 8(140.25)) For T, = 18, the ideally sampled spectrum is $,(f) = [6(fom) + uMt(fom)J. $a) 3 me a =1as” hd =, 35 9 The actual sampled spectrum is 147s(t) sine(0.U5f)[6(f-m) + pM! (f-m)] Ss) ors 07 0.63% i outs onttys estes outSta Ae *s t t f 1 | ae) 2 lO 01s =O GO 0.35 07 VO Lae (b) The ideal reconstruction filter would retain the centre 3 delta functions of S(f) or: f z — sas 0 bas With no aperture effect, the two outer delta functions would have amplitude 3. Aperture effect distorts the reconstructed signal by attenuating the high frequency portion of the message signal. Problem 3.5 The spectrum of the flat-top pulses is given by H(f) = Tsinc ( fT)exp(—jnfT) = 10™*sinc (10 f)exp(—jmf 10) Let s(f) denote the sequence of flat-top pulses: s(t) = SY m(nTh(t-n7,) ‘The spectrum Sf) = Fls(#)] is as follows: Sf) =f MUP - KH) kaw = fH) DY MUP EF) no ‘The magnitude spectrum |S(f)| is thus as shown in Fig. Ic. 1484 \rol -1/T a (a) wT f 1 (Me fs Ww f, (b) u Sif) 7 0 Wt 1/T © f VT = 10,000Hz f, = 1,000Hz W =400Hz Figure 1 149Problem 3.6 At f= 1/27, which corresponds to the highest frequency component of the message signal ae sampling. rate equal to the Nyquist rate, we find from Eq. (6-19) that the amplitude respor sine (O5777,) lea cndiion 1.0) 7 re $s 34 35 38 buy eve 17, Figure 1 of the equalizer normalized to that at zero frequency is equal to 1 (n/2XT/T) sine(O.ST/T,)_sin[(n/2NT/T,)] where the ratio T/T, is equal to the duty cycle of the sampling pulses. In Fig. 1, this result is plotted as a function of T/T,. Ideally, it should be equal to one for all values of 7/T,. For a duty cycle of 10 percent, it is equal to 1.0041. It follows therefore that for duty cycles of less than 10 percent, the aperture effect becomes negligible, and the need for equalization may be omitted altogether. 150Problem 3.7 Consider the full-load test tone A cos(2tf,t). Denoting the kth sample amplitude of this signal by A), we find that the transmitted pulse is Ay g(t), where g(t) is defined by the spectrum: co If < Bp Gf) = otherwise The mean value of the transmitted signal power is 1 its, it 2, Eins sy Ft ray g(ty iat) Lre UTS our, L 2 2 ALA, eo(t) dt) LT, ELLA S © e@tyat ba -LT, where T, is the sampling period. However, ELAAL] = o, otherusie Therefore, e(tyat Using Rayleigh's energy theorem, we may write L gtat =f peceyi2ar 151Therefore, ae The average signal power at the receiver output is A°/2. He th to-noise ratio is given by pass ness She output signal By choosing B,=1/2T,, we get (SNR)g This shows that PAM and baseband signal transmission have the same signal-to-noise ratio for the same average transmitted power, with ‘additive white Gaussian noise, and assuming the use of the minimum transmission bandwidth possible. Problem 3.8 (a) The sampling interval is T, = 125 ¥s, There are 24 channels and 1 syne pulse, so the time alloted to each channel i8 1, = T,/25 = 5 Us. The pulse duration is 1 Ms, so the time between pulses is 4 Us. (b) If sampled at the nyquist rate, 6.8 kHz, then T, = 147 vs, T, = 6.68 Us, and the time between pulses is 5.68 us. Problem 3.9 (a) The bandwidth required for each single sideband channel is 10 kHz. The total bandwidth for 12 channels is 120 kHz. (>) The Nyquist rate for each signal is 20 kHz. For 12 TDM signals, the total data rate is 2N0 kliz, By using a sinc pulse whose amplitude varies in accordance with the modulation, and with zero crossings at multiples of (1/240) ms, we need a minimum bandwidth of 120 KHz. 152Problem 3.10 (a) The Nyquist rate for s,(t) and s.(t) is 160 Hz. Therefore, at must be greater than 160, and the maximun R is 3: (b) With R= 3, we may use the following signal format to multiplex the signals s,(t) and 8,(t) into a new signal, and then multiplex s,(t) and 8,(t) and 9,(t) including markers for synchronization: Macker s eA poytid Time SS 535 535 Se SSS SSS S, @ zero sampas Based on this signal format, we may develop the following multiplexing system: 2400 Hz. Clock, FL - is Delay Dato baencett [serete} ae samt fe 5,ce (e) M [828 [e poten] v x Mutbixploud [signa xcz a Sampler, 153Problem 3.11 In general, a line code can be represented as N s(t) = YS a,g(t—n,) na Let g(t) = G(f). We may then define the Fourier transform of s(t) as N SN = DY a,GNe nN -jont, a jon’ =a) Sage nN where «= 2zf. The power spectral density of s(¢) is See | n= SA) = lim [How 5 Ii T+ a ae i(m=n)oT, =IG()P lim # Y Flajagle” ' ro n= om=N where T is the duration of the binary data sequence, and E denotes the statistical expectation operator. Define the autocorrelation of the binary data sequence as R(k) = Ela,a,.,) Indy + By letting m =n +k and T= (2N + 1)T,, we may write ig? SA) = I6(Al inde ; No keNen ar 8,7) = GDP im I T, N-=| 2N41 154- IeanP oly “T Zwe @ where 1 RU) = Eldan gl = Sanne .dP; @ in where p; is the probability of getting the product (dy dn4,); and there are J possible values for the 4, dng product. G(/) is the spectrum of the pulse-shaping signal for representing a digital symbol. Eqs. (1) and (2) provide the basis for evaluating the spectra of the specified line codes. (a) Unipolar NRZ signaling For rectangular NRZ pulse shapes, the Fourier-transform pair is a= Aree x) FG(f) = AT,sinc(fT,) For unipolar NRZ signaling, the possible levels for a’s are +4 and 0. For equiprobable symbols, we have the following autocorrelation values: 20) = tare to =A/2 4 RU) = SnAg 2):Pi ia 2 2 Aan OMOMOEAg = Fafa de Ga 4 for dl>o Thus Rk) =f A fork = 0 GB) 74 for k#0Therefore, the power spectral density for unipolar NRZ signals, using formulas (1) and (3), is 3e i ae som t ral TA 4 is But, where 8(f) is a delta function in the frequency domain. Hence, AT, 2 SCA) = sino Ty) ia] We also note that sinc(f7,) = 0 at f = +,n#0; we thus get AT, sue teine(sTy)[ +5 (b) Polat Non-return-to-zero Signaling For polar NRZ signaling, the possible values for a’s are +A and -A, Assuming equiprobable symbols, we have RO) = Say4,),P; ist For k #0, we have 156‘ RE) = YA ntg 0):Pi i=l 5 CA) 5 =0 Thus, R(k) = | A for k= 0 4) 0 for k#0 ‘The power spectral density for this case, using formulas (1) and (4), is S(f) = A’T,sine*(fT,) (c) Return-to-zero Signaling The pulse shape used for return-to-zero signaling is given by (x ) We therefore have T,/2 GY) spPsine(f1,/2) The autocorrelation for this case is the same as that for unipolar NRZ signaling. Therefore, the power spectral density of RZ signals is 2 aT, Sf) = sine’ gancTall+ (d) Bipolar Signals The permitted values of level a for bipolar signals are +A, -A, and 0, where binary symbol 1 is represented alternately by +4 and -A, and binary 0 is represented by level zero. We thus have the following autocorrelation function values:Fork>1, RW) = Day. Thus, as for k = ae AS for (a | ° 0 for |k| > 1 ‘The pulse duration for this case is equal to T,/2. Hence, Gin = Bai ine 22) © Using Equations (1), (5) and (6), the power spectral density of bipolar signals is jot, -joT,, be | sine ZA 1 -cos(2n/T,)] 158(ec) Manchester Code, The permitted values of a’s in the Manchester code are +A and -A. Hence, latte ata lta leg? R(O) = GAP HAY + GCAY? + 34?) =a For k#0, 4 RE) = Daya, dis = =0 Thus, 2 mua 4 for k = 0 0 for k#0 U The pulse shape of Manchester signaling is given by tT y/4 (t-T,/4y Hn = 0S) (FH) The pulse spectrum is therefore Gf) fT) ~ioty/4 sine) 3 srind p22) 159Therefore, the power spectral density of Manchester NRZ has the form Problem 3.12 Power spectral density of a binary data stream will not be affected by the use of differential encoding. The reason for this statement is that differential encoding uses the same pulse shaping functions as ordinary encoding methods. If the number of bits is high, then the probability of a symbol one and symbol zero are the same for both cases, Problem 3.13 (a) | (mt) Ty oT (b) an = ): 0, otherwise Equivalently, we may write a(t) = cos( FF Aree) where rect(®) is @ rectangular function of unit amplitude and unit duration. The Fourier transform of g(t) is given by Gf) = FS r-A rez] sinc(fT,) where A denotes the pulse amplitude and * denotes convolution in the frequency domain. 160Using the replication property of the delta function 8(/), we get + sine (rls + asine(n(/-F)) sine (7 or J 7) a 2 Note that the two spectral components sine (Ti f 2) and sine (rf fe 7) overlap in 0 the frequency interval -(1/T,) < f < (I/T;), hence the presence of cross-product terms in Eq. (). Figure 1 plots the normalized power spectral density S()/(A77)/4) versus the normalized frequency fT), The interesting point to note in this figure is the significant reduction in the power spectrum of the puls shaped data stream x(t) in the interval -L/T, << Ty (©) The power spectral density of the standard form of polar NRZ signaling is S(f) = AT ysine*(fT,) Comparing this expression with that of Eq. (1), we observe the following differences: SS Polar NRZ ing | Polar NRZ signals using cosine pulses rectangular pulses feo 0 “7, f= 22/T, ATy4 0 161 @)rk 162 14 12h = = © wt ws” oatProblem 3.14 (4 Ch a TELE = FR Ppl ps © HILAL 163roblem 3.15(a) PrProblem 3.15(b) cay" dn tt boo tot 165Problem 3.16 ‘The minimum number of bits per sample is 7 for a signal-to-quantization noise ratio of 40 dB. Hence, (™* number of sampl in a duration of 10s ) = 8000 10 8x10" samples The minimum storage is therefore =7x8x 10* = 5.6.x 105 = 560 kbits 166Problem 3.17 Suppose that baseband signal m(t) is modeled as the sample function of a Gaussian random process of zero mean, and that the amplitude range of m(t) at the quantizer input extends from 4Armg t0 4Arm,: We find that samples of the signal m(t) will fall outside the amplitude range 8Aymg with a probability of overload that is less than 1 in 10* If we further assume the use of a binary code with each code word having a length n, so that the number of quantizing levels is 2", we find that the resulting quantizer step size is 3 = SAnms @ aR Substituting Eq. (1) to the formula for the output signal-to-quantization noise ratio, we get (SNR), = son @ Expressing the signal-to-noise ratio in decibels: 10logg(SNR) = GR - 7.2 (3) This formula states that each bit in the code word of a PCM system contributes 6dB to the signal- to-noise ratio. It gives a good description of the noise performance of a PCM system, provided that the following conditions are satisfied: 1, The system operates with an average signal power above the error threshold, so that the effect of transmission noise is made negligible, and performance is thereby limited essentially by quantizing noise alone. 2. ‘The quantizing error is uniformly distributed. 3. The quantization is fine enough (say R > 6) to prevent signal-correlated patterns in the quantizing error waveform. 4. The quantizer is aligned with the amplitude range from -4Armg 0 4Armg- In general, conditions (1) through (3) are true of toll quality voice signals. However, when demands on voice quality are not severe, we may use a coarse quantizer corresponding to R < 6. In such a case, degradation in system performance is reflected not only by a lower signal-to-noise ratio, but also by an undesirable presence of signal-dependent patterns in the waveform of quantizing error. 167Problem 3.18 (a) Let the message bandwidth be W. Then, sampling the message signal at its Nyquist rate, and using an R-bit code to represent each sample of the message signal, we find that the bit duration is le * aR The bit rate is 1 wt BR % ‘The maximum value of message bandwidth is therefore 10° 7 7) Ynax = “2 x x = 3.57 x 10° te (b) The output signal-to-quantizing noise ratio is given by (see Example 2): 10 10819 (SNR) = 1.8 + OR 1.8467 43,8 dB Problem 3.19 Let a signal amplitude lying in the range 1 1 HO sr iy tg hy be represented by the quantized amplitude x,. The instantaneous square value of the error is (exy)*, Let the probability density function of the input signal be f(x). If the step size 6; is small in relation to the input signal excursion, then fy(x) varies little within the quantum step and may be approximated by fy(x. )+ Then, the mean-square value of the error due to signals falling within this quantum is ? tyra 168a (2) Therefore, eliminating fy(x,) between Eqs. (1) and (2), we get 2,22 EIQf] = + The total mean-square value of the quantizing error is the sum of that contributed by each of the several quanta. Hence, 169Problem 3.20 (a) = = Voltage Quaticn output Quadkzer eutpok Voltane Tyla. a Tone es ey it 83 8% * . Quoted one anpor (>) 10Problem 3 The quantizer has the following input-output curve: Ouspur At the sampling instants we havet t m(t) code 3/8 2 0011 -1/8 v2 0011 1/8 3v2 1100 43/8 3v2 1100 And the coded waveform is (assuming on-off signaling): a pouty Time (Seconds) + 3 8 7 roblem 3.2: The transmitted code words are: t/t, code 1 001 010 3 on 4 100 5 101 6 110 mtThe sampled analog signal is Problem (a) The probability p, of any binary symbol being inverted by transmission through the system is usually quite small, so that the probability of error after n regenerations in the system is very nearly equal ton p,. For very large n, the probability of more than one inversion must be taken into account. Let p, denote the probability that a binary symbol is in error after transmission through the Complete system. ‘Then, p, is also the probability of an odd mumber of errors, since an even number of errors"restores the original value. Counting zero as an even number, the probability of an even number of errors is 1-p,. Hence Pp (1-P4)+ (1-1 Poet Py C2 PLHP, This is a linear difference equation of the first order. Its solution is 1 a Py =e (1-1-2, (bv) If p, is very small and n is not too large, then (1-2) 9" = 1-2) yn and 1 Py = gli-(-2p,n)] = pyr 172Problem 3.24 - Regenerative repeater for PCM Three basic functions are performed by regenerative repeaters: equalization, timing and decision- making. Equalization: The equalizer shapes the incoming pulses so as to compensate for the effects of amplitude and phase distortion produced by the imperfect transmission characteristics of the channel. ‘Timing: The timing circuitry provides a periodic pulse train, derived from the received pulses, for sampling the equalized pulses at the instants of time where the signal-to-noise ratio is maximum. Decision-making: The extracted samples are compared to a predetermined threshold to make decisions. In each bit interval, a decision is made whether the received symbol is 1 or 0 on the basis of whether the threshold is exceeded or not. Problem 3.25 m(1) = Atanh(Br) To avoid slope overload, we require |dm(1)| ml | w am) = 4 Bsech?(Br) Q) dt Hence, using Eq. (2) in (1): A> max(ABsech*(Br)) xT, @) Since sech(Br) = —1. cosh (Bi) Bi it follows that the maximum value of sech(fr) is 1, which occurs at time 1 = 0, Hence, from Eq. (3) we find that A > ABT, 173Problem 3.26 The modulating wave is mt) = A, cos(2nf,t) The slope of m(t) is an(t) at W2nf hy sin(2nft) The maximum slope of m(t) is equal to 2nf,A, The maximum average slope of the approximating signal m,(t) produced by the delta modulator is 6/T,, where 6 is the step size and T, is the sampling period. The limiting value of 4, is therefore given by é anf, > S mn ? TS or 6 A> sq in? Bef, Ty Assuming a load of 1 ohm, the transmitted power is AC/2, Therefore, the maximum power that may be transmitted without slopeoverload distortion is equal to 6°/8x"r272, 174Problem 3.27 Is= Wxyquist Frxyquist = 6-8 KHz f= 10x 6.8 x 10° = 6.8 x 10* Hz. A 2.>max| idm(1)) dt For the sinusoidal signal m(t) = A,,Sin(2rtfyf), we have POs raf pA yC08(2%F yf) dt Hence, |dm(t)| 2: ri em = Pha or, equivalently, 4 T, Ti Arb max Therefore, Andean = oto mlmax ~ TX 20x Fy ~ AL: 2th m 01x68 x 10° 2nx 10° u 1osv 175Problem 3.28 (a) From the solution to Problem 3.27, we have a) ‘The average signal power = £ =e) With slope overload avoided, the only source of quantization of noise is granular noise Replacing A/2 for PCM with A for delta modulation, we find that the average quantization noise power is A7/3; for more details, see the solution to part (b) of Problem 3.30. The waveform of the reconstruction error (i.e., granular quantization noise) is a pattern of bipolar binary pulses characterized by (1) duration = T, = I/f,, and (2) average power = A/3. Hence, the autocorrelation function of the quantization noise is triangular in shape with a peak value of A?/3 and base 27,, as shown in Fig. 1: Rol) > v3 Fig. 1 From random process theory, we recall that SoAjao = [Rola which, for the problem at hand, yields ‘Typically, in delta modulation the sampling rate f, is very large compared to the highest frequency component of the original message signal. We may therefore approximate the power spectral density of the granular quantization noise as 1762 Soif=) A3F, “WS SSW 0, otherwise where W is the bandwidth of the reconstruction filter at the demodulator output. Hence, the average quantization noise power is 20°W 3f5 w N= J So(fdf = Q) -w Substituting Eq. (2) into (1), we get of 2 LmAY? W wa (Say _ 8x fw af (b) Correspondingly, output signal-to-noise ratio is (3) SNR = —— >, (81° f,A°W)/3f? ah lor Ww Problem 3.29 @as dhs 2 hn 172xmx 10x af 0) (SNR) = SS ax’ fw 3 fe e3e (50x13) 16m? 10°x5x 10° = 475 In decibels, (SNR) oq, = LOlog 49475 = 268 4B Problem 3.30 (@) For linear delta modulation, the maximum amplitude of a sinusoidal test signal that can be used without slope-overload distortion is Af, A= aap, 0.1.x 60x 10° 3 —veo ff, = 2x3x10° 2nx 1x10 = 0.95V (b) i) Under the pre-filtered condition, it is reasonable to assume that the granular quantization noise is uniformly distributed between -A and +A. Hence, the variance of the quantization noise is 178(SNR), prefiltered = _ 3x 0.95? Sieean ik 2x0. = 135 = 21.3 dB (ii)The signal-to-noise ratio under the post-filtered condition is 3 (3) =F (Weestneea ” Ten? fa (60 (1y’x3 = 1367 = 313 dB ‘The filtering gain in signal-to-noise ratio due to the use of a reconstruction filter at the demodulator output is therefore 31.3 - 21,3 = 10 dB. 179Problem 3.31 Let the sinusoidal signal m(t) = Asinaog, where @p = 2xfo The autocorrelation of the signal is 2 R(t) = mn c0s(Wot) A 2a _ ( 1 R,(L) = 5 °°5( %0%* ia) 2 a = F.e0s(0.1) For this problem, we thus have Ry, = (Ry, (O)) (R,(1)] (a) The optimum solution is given by Wo = Rie, 2 Fos (0.1) = 2 = 00s(0.1) ae 2 = 0.995 (©) Jig = Ry(0) = TRG Ey 2 2 2 5 4-4 .08(0.1) x 4-cos(0.1)/(A?/2) 180(1 cos?(0.1)) np = 0.00547 Problem 3.32 1 08 06] = |08 1 08) 0.608 1 = [os, 06, 04)” (a) Wo = RY -1 1 08 06] [os = |08 1 0.8} 0.6) 0.608 1} [o4| 0.875 =l 0 -0.125 (©) Jin = R,(0)— 4, Ri 'r, r, = R,(O0)—1 Wo 0.875 08,06, 0.4]| 0 —0.125] (0.8 x0.875- 04x 0.125) 0.7 + 0.05 0.35 181Problem 3.33 RK =|! 08 08 1 (@) wo = Rr = | 0.8889 -O.1LIL () Imin = R,(O)— ERY 10.6444 = 0.3556 which is slightly worse than the result obtained with a linear predictor using three unit delays (ic., three coefficients). This result is intuitively satisfying. Problem 3.34 Input signal variance = R,(0) The normalized autocorrelation of the input signal for a lag of one sample interval is, = 0.75 Error variance = R,(0)-R,(1)Rz!(0)R,(1) = R,(0)(1-p2(1)) 182RO) Processing gain. = ———*—__ RO) ~P-(1)) 2.2857 Expressing the processing gain in dB, we have 3.59 dB 10log 19(2.2857) Problem 3.35 Processing gain (a) Three-tap predictor: Processing gain = 2.8571 4.56 dB (b) Two-tap predictor: Processing gain = 2.8715 4.49 dB Therefore, the use of a three-tap predictor in the DPCM system results an improvement of 4.56 - 4.49 = 0.07 dB over the corresponding system using a two-tap predictor. Problem 3. (a) For DPCM, we have 10log9(SNR)o = 01 + 6n dB For PCM, we have 10logyg(SNR)g = 4.77 + 6n - 20log;o(log(1 + 11)) where n is the number of quantization levels SNR of DPCMSNR = ce + 6n, where -3<0< 15 For n=8, the SNR is in the range of 45 to 63 dBs. SNR of Pt SNR = 4.77 + 6n - 20logyo(log(2.56)) 4.77 +48 - 14.8783 =38dB Therefore, the SNR improvement resulting from the use of DPCM is in the range of 7 to 25, 4B. (b) Let us assume that m) bits/sample are used for DPCM and » bits/sample for PCM If = 15 dB, then we have 15 +6m, = 6n- 10.0 10+15, 6 Rearranging: (n ~ nj) = 4.18 which, in effect, represents a saving of about 4 bits/sample due to the use of DPCM. If, on the other hand, we choose -3 dB, we have -3 +6n, = 6n - 10 Rearranging: (n-n,) = 12-3 = 101 which represents a saving of about 1 bit/sample due to the use of DPCM. 184,Problem 3.37 The transmitting prediction filter operates on exact samples of the signal, whereas the receiving prediction filter operates on quantized samples. Problem 3.38 Matlab codes 4% Problem 3.38, CS: Haykin Ylat-topped PAM signal Yand magnitude spectrum o Mathini Sellathurai Yaata £3=8000; % sample frequency ts=1.260-4; Yit/ts pulse_duration=5e-S; Ypulse duration % sinusoidal sgnai; 260-6; Ysampling frequency of signal £4=80000; t= (0:td: 100¥td); m=10000; sesin(fmet); % PAM signal generation pan_s=PAM(s,td, ts, pulse figure(1);hold on 185plot(t,s,’—-"); plot(t(1:length(pam.s)),pam_s); xlabel('tine’) ylabel (‘magnitude’) Legend ‘signal’, *PAN-signal’); % Computing magnitude spectrum S(#) of the signal a=((abs(t2t(pam.s)).-2)); a=a/nax(a); f=t50(ts/24:ts%(t5/ta) : CLength(a))+£a¢ (£2/24); figure (2) plot(t,a); xlabel(’frequency’); ylabel (‘magnitude’) 4% finding the zeros index=tind(a ost > Output 1 Fitter matched > to 59(0) > Output 2 — input Filter matched - to 54(0) Output Fig. 4 202Problem 4.3 Ideal low-pass filter with variable bandwidth. The transfer function of the matched filter for a rectangular pulse of duration t and amplitude A is given by Hopt(®) = sine(fT)exp(-jnfT) @ The amplitude response |H,,,(f)| of the matched filter is plotted in Fig. 1(a). We wish to approximate this amplitude response with an ideal low-pass filter of bandwidth B. The amplitude response of this approximating filter is shown in Fig. 1(b). The requirement is to determine the particular value of bandwidth B that will provide the best approximation to the matched filter. We recall that the maximum value of the output signal, produced by an ideal low-pass filter in response to the rectangular pulse occurs at t = T/2 for BT < 1. This maximum value, expressed in terms of the sine integral, is equal to (2A/x)Si(nBT). The average noise power at the output of the ideal low-pass filter is equal to BNy. The maximum output signal-to-noise ratio of the ideal low-pass filter is therefore = (2A/n)?Si2(nBT) (2) (SNR), a 0 Thus, using Eqs. (1) and (2), and assuming that AT = 1, we get (SNR, 2 _2_ 5 2¢¢8T) GNR, BT This ratio is plotted in Fig. 2 as a function of the time-bandwidth product BT. The peak value on this curve occurs for BT = 0.685, for which we find that the maximum signal-to-noise ratio of the ideal low-pass filter is 0.84 dB below that of the true matched filter. Therefore, the " best” value for the bandwidth of the ideal low-pass filter characteristic of Fig. 1(b) is B = 0.685/T. 203Hope (fl 10 ae 2 a r r (a) a a \H(f)l 1.0 f -B O B (b) Figure 1 10 Matched filter eos ores hie s\e B/S os ee a 0.2 Figure 2 204Problem 4.4 output of the low-pass RC filter, produced by a rectangular pulse of amplitude A and duration T, is as shown below: Soft) A(L-exp(-2nfT)) | — 0 T t The peak value of the output pulse power is. Poy = ALL ~ exp(-20 fT) where fy is the 3-dB cutoff frequency of the RC filter. The average output noise power is No Nout = om — Noto “8 The corresponding value of the output signal-to-noise ratio is therefore Nenfglt ~ HP2RF7I Differentiating (SNR)p with respect to foT and setting the result equal to zero, we find that (SNR)ouy attains its maximum value at 0.2 fo ‘The comesponding maximum value of (SNR)oyx isol - exp(-0.40) 7 For a perfect matched filter, the output signal-to-noise ratio is 2E (SNR)o matched = No os No Hence, we find that the transmitted energy must be increased by the ratio 2/1.62, that is, by 0.92 4B so that the low-pass RC filter with fy = 0.2/T realizes the same performance as a perfectly matched filter. Problem 4.5 ©) Po>Pr The transmitted symbol is more likely to be 0. Hence, the average probability of symbol error is smaller when a 0 is transmitted than when a 1 is transmitted, In such a situation, the threshold A in Figs. 4.5(a) and (b) in the textbook is moved to the right. Gi) Py > Po ‘The transmitted symbol is more likely to be 1. Hence, the average probability of symbol error is smaller when a 1 is transmitted than when a 0 is transmitted. In this second situation, the threshold din Figs. 4.5(a) and (b) in the textbook is moved to the left. 206Problem 4.6 ‘The average probability of error is Pe = pr {% fyly Idx + po f° fyty lnndx @ An optimum choice of ) corresponds to minimum P.,. Differentiating Eq. (1) with respect to A, we get: ap, x = py fy IL) - pofy lo) ap, Setting —* = 0, wo got the following condition for the optimum value off: £yQope I) _ Po FyMopt 10) Pi which is the desired result. 207Problem 4.7 Ina binary PCM system, with NRZ signaling, the average probability of error is P, = d erk fe 2 0 The signal energy per bit is Ey = A?T) where A is the pulse amplitude and 7, is the bit (pulse) duration. If the signaling rate is doubled, the bit duration Ty, is reduced by half. Correspondingly, Ey, is reduced by half. Let u = ¥E,/No. We may then set P, = 10-6 = 3 erfe(u) Solving for u, we get u=33 When the signaling rate is doubled, the new value of P, is P, = 2 erfe| & og ote = 2 erte(2.33) 2 = 10°. 208where Ey, = AT. We may rewrite this formula as, P, = 1 erfe(A @ 2 o where A is the pulse amplitude at o = VNoT,. We may view 0” as playing the role of noise variance at the decision device input. Let al> We are given that o? = 10 volts, 6 = 0.1 volt P, = 10° Since P, is quite small, we may approximate it as follows: erfe(u) = 2xe(-u?) vru‘We may thus rewrite Eq. (1) as (with P, = 10%) en Nir = 108 Solving this equation for u, we get u = 3.97 ‘The corresponding value of the pulse amplitude is A=ou=0.1x 3.97 = 0.397volts (b) Let 6%, denote the combined variance due to noise and interference; that is 2 tag? Baotedl where o” is due to noise and 6%, is due to the interference. The new value of the average probability of error is 10°. Hence 106 = t we(] 2 oT (2) = + erfe(u) where een Or 210Equation (2) may be approximated as (with P, = 10°) exp(-up) 2a up 106 Solving for up, we get up = 3.37 ‘The corresponding value of 02, is 2 o8 = ‘AN. (0-397 F ~ 0.0138 volts? uy) (3.37 ‘The variance of the interference is therefore 2 of = op - 0? = 0.0138 - 0.01 = 0.0038 volts”Problem 4.9 Consider the performance of a binary PCM system in the presence of channel noise; the receiver is depicted in Fig. 1. We do so by evaluating the average probability of error for such a system under the following assumptions: 1. The PCM system uses an on-off format, in which symbol 1 is represented by A volts and symbol 0 by zero volt. 2, The symbols 1 and 0 occur with equal probability. 3. The channel noise w(t) is white and Gaussian with zero mean and power spectral density Ny2. To determine the average probability of error, we consider the two possible kinds of error separately. We begin by considering the first kind of error that occurs when symbol 0 is sent and the receiver chooses symbol 1. In this case, the probability of error is just the probability that the correlator output in Fig. 1 will exceed the threshold 4 owing to the presence of noise, so the transmitted symbol 0 is mistaken for symbol 1. Since the a priori probabilities of symbols 1 and O are equal, we havep, = p, Correspondingly, the expression for the threshold A simplifies as follows: raat @ 2 where TT, is the bit duration, and A", is the signal energy consumed in the transmission of symbol 1. Let y denote the correlator output: y = £7 stoxtat (2) Under hypothesis Ho, corresponding to the transmission of symbol 0, the received signal x(t) equals the channel noise w(t). Under this hypothesis we may therefore describe the correlator output as Hoy =A ie w(t)dt @) Since the white noise w(t) has zero mean, the correlator output under hypothesis Hy also has zero mean, In such a situation, we speak of a conditional mean, which (for the situation at hand) we Aladescribe by writing No = ELY [Hol = 5| ic wood | Co) where the random variable Y represents the correlator output with y as its sample value and W(t) is a white-noise process with wit) as its sample function. The subscript 0 in the conditional mean Hp tefers to the condition that hypothesis Hy is true. Correspondingly, let oy denote the conditional variance of the correlator output, given that hypothesis Hy is true. We may therefore write 08 = ELY? [Hol 6) : 3| G7 L7 wenwepraty dt, The double integration in Eq. (5) accounts for the squaring of the correlator output. Interchanging the order of integration and expectation in Eq. (5), we may write of = £7 [7 EIWit)Wty)ldty dt, © = fT Ch Ruts - teddy dt The parameter Ry(ty - t,) is the ensemble-averaged autocorrelation function of the white-noise process W(t). From random process theory, it is recognized that the autocorrelation function and power spectral density of a random process form a Fourier transform pair. Since the white-noise process W(t) is assumed to have a constant power spectral density of Ny/2, it follows that the autocorrelation function of such a process consists of a delta function weighted by No/2. Specifically, we may write Ry - ) = Mae - 4 +) @ Substituting Eq. (7) in (6), and using the property that the total area under the Dirac delta function &(¢ - t, + t,) is unity, we got 2 of = NoToA ® 2 2 213The statistical characterization of the correlator output is compizted by noting that it is Gaussian distributed, since the white noise at the correlator input is itself Gaussian (by assumption). In summary, we may state that under hypothesis Hy the correlator output is a Gaussian random variable with zero mean and variance NgT}A7/2, as shown by fly) = 2 exp| - =, (9) IRNoT, A NoTpA where the subscript in f,(y) signifies the condition that symbol 0 was sent. Figure 2(a) shows the bell-shaped curve for the probability density function of the correlator output, given that symbol 0 was transmitted. The probability of the receiver deciding in favor of symbol 1 is given by the area shown shaded in Fig. 2(a). The part of the y-axis covered by this area corresponds to the condition that the correlator output y is in excess of the threshold A defined by Eq, (1). Let Pyg denote the conditional probability of error, given that symbol 0 was sent. Hence, we may write Pro = f° fo) ay 1g A (10) 7 y = exp| - —>—_. |dy |mNoTy a 7? { NoTA? Define ze an We may then rewrite Eq. (10) in terms of the new variable z as 1 fe 2) Pio = exp(-2?) da (a2) vn “VAAN which may be reformulated in terms of 214complementary error function 2 fe 2 erfe(u) = — exp(-z7) dz (13) Ie 5 Accordingly, we may redefine the conditional probability of error Pg os A*T, “AN (4) 1 =f ert dapat Consider next the second kind of error that occurs when symbol 1 is sent and the receiver chooses symbol 0. Under this condition, corresponding to hypothesis H,, the correlator input consists of a rectangular pulse of amplitude A and duration Ty, plus the channel noise w(t). We may thus apply Eq. (2) to write Hy:y=A ( [A + wit)] dt (15) ‘The fixed quantity A in the integrand of E q. (15) serves to shift the correlator output from a mean value of zero volt under hypothesis Hy to a mean value of A*T,, under hypothesis H,. However, the conditional variance of the correlator output under hypothesis H, has the same value as that under hypothesis Hy. Moreover, the correlator output is Gaussian distributed as before. In summary, the correlator output under hypothesis Hy is a Gaussian random variable with mean AD, and variance NoT},2/2, as depicted in Fig. 2(b), which corresponds to those values of the correlator output less than the threshold A set at A*I\/2, From the symmetric nature of the Gaussian density function, it is clear that For * Be ae Note that this statement is only true when the a priori probabilities of binary symbols 0 and 1 are equal; this assumption was made in calculating the threshold 2. To determine the average probability of error of the PCM receiver, we note that the two possible kinds of error just considered are mutually exclusive events. Thus, with the a priori probability of transmitting a 0 equal to p, and the a priori probability of transmitting a 1 equal top ,we find 215that the average probability of error, P,, is given by + . 17) PIO” PP “ Since 7, = pyg and p,+p, = 1, Ha. (17) simplifies as Pe = Pro = Poy or A’, 1 P, = A erfe| 1 b as) 2 24 No Choose H, it 5 Nisexcedded xy —)—H] Jy ae L-»| Decision} g Otherwise, t choose Hy s(t) » Figure 1 foo! Bo 0 5a ° Ao 1 i i i ' 7 An, * Figure 2 216Problem 4.10 For unipolar RZ signaling, we have Binary symbol 1: s(t) = 4A for 0 <1< 712 and s(t) =0 for 7/2 A, as shown byP(error]0) = J” Fyr|0ay We then have P(error|0) = Define z = and so write 1 2 P y=4f -2")de (error| 1) Flim exp(-z") NoP(error|1) The average probability of error is therefore P(1)P(error|1) + P(error|0)P(0) 1 = ert 5 E,7N,) z Leste 1 [4 | pees aK, | w The average probability of error for on-off (i.e., unipolar NRZ) type of encoded signals is dosed AT) B34 Ne | Comparing this result with that of Eq. (1) for the unipolar RZ type of encoded signals, we immediately see that, for a prescribed noise spectral density No, the symbol energy in unipolar RZ. signaling has to be doubled in order to achieve the same average probability of error as in unipolar NRZ signaling. Problem 4.11 Probability of error for bipolar NRZ signal A Binary symbol 1 : s(t) Binary symbol 0: s(¢) = 0 Energy of symbol 1 = E, = A77), 219‘The absolute value of the threshold is A= Refering to Fig. 1 on the next page, we may write P(errorls exp| [* + a ie = af{5 2) «(3 | P(error|s = -A) Similarly, P(error|s = +A) P(error|s = 0) = Fehon Fa (1 [Eo «aiff The average probability of error is therefore P P(s=+A)P(error|s= +A) + P(s=0)P(error|s = 0) The conditional probability density functions of symbols 1 and 0 are given in Fig. 1:Fuly|s=0) pr _| Srols=+A) WE, Figure 1 224Problem 4.12 ‘The rectangular pulse given in Fig, P4.12 is defined by (0) = rec(WT) The Fourier transform of g(t) is given by 1/2 GA =f pa SPCP2np dt = Tsinc (fT) We thus have the Fourier-transform pair rec(t/T) # T sine (fT) ‘The magnitude spectrum |G(f/T is plotted as the solid line in Fig. 1, shown on the next page. Consider next a Nyquist pulse (raised cosine pulse with a rolloff factor of zero). The magnitude spectrum of this second pulse is a rectangular function of frequency, as shown by the dashed curve in Fig. 1. Comparing the two spectral characteristics of Fig. 1, we may say that the rectangular pulse of Fig. 4,12 provides a crude approximation to the Nyquist pulse. 2221 T 7 | oof ggeist poles Wil, zaco vallefe f j~— { 1 ost o7 ler | os oa a2 oz o4 i) -¥T ~2/r Vem hr 9 ag VE ar afr Frequna , f Figen I Spechel chancheei shies 223Problem 4.13 Since P(f) is an even real function, its inverse Fourier transform equals pit) = 2 [ PCD cost2ntt) df @ The P(f) is itself defined by Eq. (7.60) which is reproduced here in the form w o< kl<& : f @ Pe af vo Fl- If. f 2w-f, Hence, using Eq. (2) in (1): -1 fh Ne cos MED Pit) = = Gh coscanty af ae, ht one | feosenm or , [sinenty PY 4nwWt |, ew-f, : 5) PW sinlonty + 74D sinfantt - 75-5. + lw 2Wo. eee awe te ant + W2Wa | 4w Qnt — W2Wo I = Sin(2nft) sin[2xt(2W-f,)] Fewt 4nWt 1 sin(2nfyt) + sinf2n2W-f)] | sin(@nfyt) + sinl2nt(2W-f,t)] aw Ont — mWa 2nt — W2Wo 1). . 1 = 2 [sin(antyt) + sintam2w-fJ] | - ___™ __. Milman »)] ; * =| (ant)? 224223 = 7 fein(enwe)eos(2naw)) |——~ @2Wo? __ w Ant [(2nt)? - (W/2Wa)* i 1 = sinc(2Wt) cos(2naWt) | —__-__ 16 o?W? al Problem 4.14 The minimum bandwidth, Br, is equal to 1/2T, where Tis the pulse duration. For 64 quantization levels, log,64 = 6 bits are required. Problem 4.15, The effect of a linear phase response in the channel is simply to introduce a constant delay + into the pulse p(t). The delay t is defined as -1/2n)times the slope of the phase response; see Eq. 2.171. 225 WyProblem 4.16 The Bandwidth B of a raised cosine pulse spectrun is 2W - where W vat, and W(1-a), Thus B= W(t+x), For a data rate of 56 kilobits per second, W = 28 kz. (a) For a= 0.25, B= 28 kHz x 1.25 = 35 kHz (b) 28 kz x 1.5 = 42 kHz (eo) B= 49 kHz (a) B= 56 kiz Problem 4.17 The use of eight amplitude levels ensures that 3 bits can be transmitted per pulse. The symbol period can be increased by a factor of 3. All four bandwidths in problem 7./2 will be reduced to 1/3 of their binary PAM values. Problem 4.18 (a) For a unity rolloff, raised cosine pulse spectrum, the bandwidth B equals 1/T, where T is the pulse length. Therefore, T in this case is 1/12kHz. Quarternary PAM ensures 2 bits per pulse, so the rate of information is 2M , 2y kxiooits per seoond, -(b) For 128 quantizing levels, 7 bits are required to transmit an amplitude. The additional bit for synchronization makes each code word 8 bits. The signal is transmitted at 2u kilobits/s, so it must be sampled at 2u_kbits/s Bvits/sanpie ~ 3 KHz. The maximum possible value for the signal's highest frequency component is 1.5 kHz, in order to avoid aliasing. 226Problem 4.19 The raised cosine pulse bandwidth B= 2W - f,, where W/= 1/2T,, For this channel, B= 75 kHz. For the given bit duration, W= 50 kiiz, Then, f,22W-B = 25 kz = 1- t,/8, 20.5 Problem 4.20 The duobinary technique has correlated digits, while the other two methods have independent digits. : 227Problem 4.2 (a) binary sequence by o Oo 1 1 oO 1 oo 1 omen ee a duobinary coder output % eee ete s ee ee output binary sequence o Oo 1 1 O 1 oo 1 (b) receiver input Sere eee een oO receiver output 1 V1 1-1 1-1-1 1 output binary sequence 0 1 oO 1 O 1 oo 1 We see that not only is the second digit in error, but also the error propagates. Problem 4.22 (a) binary sequence by o 0 1 1 0 1 0 0 1 coded sequence dj 1 ett Ont leet) O10] 0) a polar representation se duobinary coder output o, 2 2 0 0 2 0 -2 -2 0 receiver output o 0 1 41 0 41 0 0 14 (b) receiver input 20 00 2 0 ° receiver output o 1 4 41 0 1 001 In this case we see that only the second digit is in error, and there is no error Propagation. Problem 4.23 (a) The correlative coder has output Fn =n 7 Ynat ItS impulse response is otherwise. The frequency response is 228H(t) = hy exp(—J2nfkT,) z kz: = 1 = exp(-J20fT,) (>) Let the input to the differential encoder be x,, the input to the correlative coder be qs and the output of the correlative coder be z, Then, for the sequence 010001101 in its on-off form, we have Then z, has the following waveform The sequence z, is a bipolar representation of the input sequence x,+ Problem 4. (a) The output symbols of the modulo-2 adder are independent because: ie 2. the input sequence to the adder has independent symbols, and therefore knowing the previous value of the adder does not improve prediction of the present value, i.e. feyatyy 4) = £0q) > where y, is the value of the adder output at time nT,» The adder output sequence is another on-off binary wave with independent symbols. Such a wave has the power spectral density (from problem 4/0) , a 2, oes Sy(f) = Of) + sino*(rT,) . 229The correlative coder has the transfer function Hf) = 1 = exp(-J2xfT,)» Hence, the output wave has the power spectral density S7(f) mice? sy (1 = exp(-g2ee7,)] (1 - exp(jentT, I sy(0) [2 - 2 cos(2mfT,)] Sy(f) 2, 4 sin®(xtT,) SYP) 2, 2 AT, 4 sin®(net,y CA ace) + a? sino*(eT,)1 2, 2, ‘Ty sin*(xfT,) sinc“(fT,) In the last line we have used the fact that sin(afT, Oat f= 0. (b) gp Note that the bipolar wave has no de component. (Note: The power spectral density of a bipolar signal derived in part (a) assumes the use of a pulse of full duration 7, On the other hand, the result derived for a bipolar signal in part (d) of Problem 3.11 assumes the use of a pulse of half symbol duration T;,) 230Problem 4.25 The modified duobinary receiver estimate is 4, = ¢ (a) binary sequence a, bipolar representation modified duobinary 0, receiver output 8, output binary sequence (b) receiver input receiver output output binary sequence 0 -1 “1 1 1 1 1 Here we see that not only is the third digit in error, Problem 4.26 (a) binary sequence by coded sequence a, polar representation -1 -1 modified duobinary &% receiver output b, = loyl output binary sequence (>) receiver input receiver output output binary sequence ole) <1 ot 2 <1 -1 oo 2 3-1 oo but also a) lee 1-1 oo oo 0 0 oo a) oo O @ 104 o 0 14 o 4 an) the error propagates. -1 -2 This time we find that only the third digit is in error, “propagation. 231 o 2 o 2 4 o 2 o 2 o 4 and there is no errorProblem 4.27 (a) Polar Signalling (M=2) In this case, we have t mt) = EA, sinetg - n) where A, = * A/2. Digits 0 and 1 are thus represented by -A/2 and +A/2, respectively. The Fourier transform of m(t) is MCE) t 2A, Flsine(y - n)] Trect(fT) Z A, exp(-jemer) Therefore, m(t) is passed through the ideal low-pass filter with no distortion. The noise appearing at the low-pass filter output has a variance given by N ar Suppose we transmit digit 1, Then, at the sampling instant, we obtain a random variable at the input of the decision device, defined by A XeZeNn where N denotes the contribution due to noise. The decision level is 0 volts. If x > 0, the decision device chooses symbol 1, which is a correct decision. If X < 0, it chooses symbol 0, which is in error, The probability of making an error is ° PCKCO) =F fyCx) dx The expected value of X is A/2, and its variance is o°. Hence,fyi) 2 P(x<0) xo $ ertec 2v20 Similarly, if we transmit symbol 0, an error is made when X > 0, and the probability of this error is A P(x0) = F erfet 2ve0 ) Since the symbols 1 and 0 are equally probable, we find that the average probability of error is = $ P(x<0 | transmit 1) + $ PGOO | transmit 0) 2veo (b) Polar ternary signaling In this case we have t w(t) = EA, since n) The 3 digits are defined as follows Digit Level ° “A 1 0 2 +A Suppose we transmit digit 2, which, at the input of the decision device, yields the random variable Xa Aan 233The probability density function of X is 2 1 (ay £00 = xp (= ) 0 Vimo 20% The decision levels are set at -A/2 and A/2 volts. Hence, the probability of choosing digit 1 is we 2 1 expt- UA) ax -2 Yano 20° pedcx ch =} Lerte(—4A_) ~ erto( 34} ao ao Next, the probability of choosing digit 0 is If we transmit digit 1, the random variable at the input of the decision device is X=N The probability density function of X is therefore The probability of choosing digit 2 is A wee Pa > & = d ert) The probability of choosing digit 0 is px <4 1 B erfot ave o Next, suppose we transmit digit 0. Then, the random variable at the input of the decision device is Xe-kan The probability density function of X is therefore 2342 ext Gp Yano 20° £00 = The probability of choosing digit 1 is Ay ~ erzoc—3h Pe Scx< veo av2o 2 lerfe ¢ 0] The probability of choosing digit 2 is 3A) 2720 eee PK > 5) = 5 erfet Assuming that digits 0, 1, and 2 are equally probable, the average probability of error is 1 A A 3A 1 1 3A Poze erfo(: = 5 erfeol- J1+ 4° 5 lerfe(——)] 3 avBo 2 ao 3 2 2v8 o 1.1 A 1 1 A +3 °° > lerfe(——)] + = * 5 [erfet- 1 3 e vBo «3 2 28 o + 1 1 A 3A 1 1 3A gz ° 5 lerfe(——>) - erfet- J] +5 ° 5 erfel- ) ate 2v2 o ao 3 ? 2vB 6 ao 3 wo (c) Polar quaternary signaling In this case, we have Suppose we transmit digit 3, which, at the input of the decision device, yields the random variable: 2353A xe gten The decision levels are 0, + A, The probability of choosing digit 2 is 1A —— FS expl- van o 0 POO< X A) = Lf expl-———s— ] ax Vio 8 20° = derto 4, aoa 236The probability of choosing digit 1 is 1 oO (xe am Vo =A 20 P(-A < X <0) All 2 Lertoc—4_» ~ erte(—34} ao avo The probability of choosing digit 0 is 1 A t expl- Ven oe P(X < =A) = The probability of choosing digit 0 is A 2% o P(X < A) = 5 erfet ) vie The probability of choosing digit 2 is = erfo(—34_ ] PCO < X A) =F erfo(4y , 2% o Finally, suppose we transait digit 0, obtaining 237The probability of choosing digit 1 is P(-A A) =F ertot Since all 4 digits are equally probable, with a probability of occurence equal to 1/4, we find that the average probability of error is Pog 2d tert - erro¢y 22 0 22 6 + erto(34_) ~ erre¢—2A_5 2a avo + erto(4A 22 o + erfo(—4_ 28 6 ao 2% o Problem 4.28 The average probability of error is (from the solution to Problem 7-23) a 2The received signal-to-noise ratio is (SNR) 9 That is a TAN, A. > @) wat Substituting Eq. (2) in (1), we get 3CSNR: =p errec J— 2(M%=1) with P, = 107% e we may thus write 10° 2 1 - dy ertocuy @) wnere 3(SNR) © 201) For a specified value of M, we may solve Eq. (3) for the corresponding value of u. We may thus construct the following table: 4 u 2 3.37 4 = 3682 8 3.45 16 3.46 We thus find that to a first degree of approximation, the minimum value of received signal-to-noise ratio required for P, < 1076 is given by 30S¥R)p nin tt = (3.42)? acme) That is, (SNR)p min = 7.8 (M? —1) ooProblem 4.29 ‘Typically, a cable contains many twisted pairs. Therefore, the received signal can be written as N r(n) = Siv(n)+d(n), large N where d(n) is the desired signal and} v,(7) is due to cro: ia independent and identically distributed. Hence, by using the central limit theorem, as N becomes N infinitely large, the term ¥'v,(1) is closely approximated by a Gaussian random variable for each ial alk. Typically, the v, are statistically time instant n. Problem 4.30 (a) The power spectral density of the signal generated by the NRZ transmitter is given by 2 ae S(f) = TIGA) co) where 6” is the symbol variance, T is the symbol duration, and 1 rn ~i2mft a = L sinc Gf) a PN a = Tsinc (fT) = pone(4) @ a is the Fourier transform of the generating function for NRZ symbols. Here, we have used the fact that the symbol rate R = 1/T. A 2BIQ code is a multi-level block code where each block has 2 bits and the bit rate R = 2/T (ie., m/T, where m is the number of bits in a block). Since the 2BIQ pulse has the shape of an NRZ pulse, the power spectral density of 2BIQ signals is given by 2 FG ro N where sin(2m(f/R)) G. aight) ae 240‘The factor ./2 in the denominator is introduced to make the average power of the 2BIQ signal equal to the average power of the corresponding NRZ signal. Hence, = 9 (sin(2n(f/R)))? Aanp/ = 7Z sinc*(2s7R) @) (b) The transfer functions of pulse-shaping filters for the Manchester code, modified duobinary code, and bipolar return-to-zero code are as follows: (i) Manchester code: Gf) = 4p ~ cos( nf) 4 (Modified duobinary code: Gf) = ia [cos(3nf) - cos(nf)] ©) Gii)Bipolar return-to-zero code: Gif) = [s(n Z) x sin(xf)] © Hence, using Eqs. (4), (5), and (6) in the formula of Eq, (1) for the power spectral density of PAM line codes, we get the normalized spectral plots shown in Fig. 1. In this figure, the spectral density is normalized with respect to the symbol variance o? and the frequency is normalized with respect to the data rate R. From Fig. 1, we may make the following observations: Among the four line codes displayed here, the 2BIQ code has much of its power concentrated inside the frequency band -R/2. << R/2, which is much more compact than all the other three codes: Manchester code, modified duobinary code, and bipolar return-to-zero code. 241Manchester === Modified duo-binary Bipolar RZ 7 i 2B1Q . —— Bipolar NRZ 1.8 S(Nio*Problem 4.31 The tapped-delay-line section of the adaptive filter is shown below: ata] abet) x2] xim3) pate F[n] = x" [ndwin] d{n) = x[n]+r{n] Enror signal e[n} = d[n]- Fn] Wine 1] = Win] +uxln](din] — x" [n}w{n]) where W[n] [Woln),-) %,,[2] 1 x(n] = [xl], xfn—1],+,a[2—m]")] b= leaming parameter Pr 2 @) Input Output »Outpul > Channel > [Equalizer a no The h(t) is defined by 243” a(t) = Sw, 80-7) NY The impulse response of the cascaded system is given by the convolution sum N Py = Dey PN where p, = p(T). The kth sample of the output of the cascaded system due to the input sequence {/,} is defined by Te = Pole D InPron nek where pol; is a scaled version of the desired symbol J;. The summation term J) 1,P¢_,, i nak the intersymbol interference. ‘The peak value of the interference is given by N ww DN) = YD lod = DPD een een len mo tao ‘To make the ISI equal to zero, we require a] n=0 0, #0 (b) By taking the z-transform of the convolution sum Ny Dent Jen and recalling that convolution in the discrete-time domain is transformed into multiplication in the z-domain, we get P(z) = H(z)C(z)For the zero-forcing condition, we require that P(2) = 1. Under this condition, we have H(z) = 1/C(2) which represents the transfer function of an inverse filter. If the channel contains a spectral null at f= 1/27 in its frequency response, the linear zero- forcing equalizer attempts to compensate for this null by introducing an infinite gain at frequency f = 1/27. However, the channel distortion is compensated at the expense of enhancing additive noise: With H(z) = 1/C(z), we find that when C(z) = H(z) = © which results in noise enhancement. Similarly, when the channel spectral response takes a smaller value, the equalizer will introduce a high gain at that frequency. Again, this tends to enhance the additive noise. Problem 4.33 (a) Consider Eq. (4.108) of the textbook, which is rewritten as N F[Ru-0 P8e— 2) }oce yar = 4-1) Expanding the left-hand side: Ni FF Ryle aeceyae + J P(r Decree = q(-t) Applying the Fourier transform: [P Rye-necoe| = F{R,(t-1)} x FE(c(2))} = SAC) N N elf. 5-1 cove] Scr) 245F{a(-t)} = OCS) = O*(f) In these three relations we have used the fact that convolution in the time domain corresponds to multiplication in the frequency domain. Putting these results together, we get Non SHNCN) + FCA) = O*CA) or (s+ Bon = or(/) which is the desired result. (b) The autocorrelation function of the sequence is given by RAT) = YakT,— ty )qkT,-%) € Using the fact that the autocorrelation function and power spectral density (PSD) form a Fourier transform pair, we may write PSD = F{R,(t),t)} *{Satrsspater, 2} ‘ 1 ky? “13 7) F where F{q(t)} = O(f) 246Problem 4,34 D (a) The channel output is x(t) a, s(t-ty,) +25 s(t-ty,) Taking the Fourier transform of both sides: X(£) = Lay exp(—J2ntty,) + a, exp(—J2aft,,)] Sf) The transfer function of the channel is a) xf) S(ty ay exp(-J2ntty,) +a, exp(-J2"fty,) (b) Equalizer HA (£) ‘e Ideally, the equalizer should be designed so that HAC) HCD) = Ky exp(-Janrt,) where Ky is a constant gain and t, is the transmission delay, The transfer function of the equalizer is HO(Q) = Wy + Wy exp(-J2mfT) + wy exp(—j4nfT) ail “ Mo (1 +g expl-senet) + ig OPCS) a Therefore Ky exp(-J2nft,) HOY 4D Ky exp(-Janfty) * ay PCIE) ay KPmIeTe,) 247_ Kofay) expl-s2"f(ty=ty,)] 01 a, 2 1 a expl-Janf(ty, ~ ty,)I Since a << a4, we may approximate H,(f) as follows % ae BoA) = Zo expl-s2ntty — tyy)] (1 ~ Zo expl-s2at(tyg ~ ty] a, 2 2. 5 + ra expl=Jtaf(ty - ty,)]} Comparing Eqs. (1) and (2), we deduce that » We find that the tap weights of the equalizer are as follows 248 (2)Problem 4.35 The Fourier transform of the tapped-delay-line equalizer output is defined by oul = HO Xjq(®) @ where H(f) is the equalizer’s transfer function and X,,(f) is the Fourier transform of the input signal. The input signal consists of a uniform sequence of samples, denoted by {x(nT)). We may therefore write (see Eq. 6.2): x= 25 xe @ r where T is the sampling period and s(t) is the signal from which the sequence of samples is derived. For perfect equalization, we require that You) = 1 for all f From Eqs. (1) and (2) we therefore find that T H) = — 7 _ y XW) ® k (sequenced Let the impulse responsefof the equalizer be denoted by (w,). Assuming an infinite number of taps, we have Hf) = ry Wy exp(2nfT) nsv0 We now immediately see that H(f) is in the form of a complex Fourier series with real coefficients defined by the tap weights of the equalizer. The tap-weights are themselves defined by Wat $ [2h HOexp(-j2nt/1), n=6, +1, +2... 249The transfer function H(f) is iteelf defined in terms of the input signal by Eq, (3). Accordingly, a. tapped-delay-line equalizer of infinite length can approximate any function in the frequency interval (-/2T, V/2T). Problem 4.36 (a) Asan example, consider the following single-parameter model of a noisy system: d{n] = wolnlxin] + ln) where x[v] is the input signal and v{n] is additive noise. To track variations in the parameter woln], we may use the LMS algorithm, which is described by Error signal n) stn (d| w[n]x[n]) = (1- px" [nwfn] + wxlnld{n] a) To simplify matters, we assume that W{n] is independent of x(n]. Hence, taking the expectation of both sides of Eq. (1): El[n+ 1] = (1 poeta] + wry. 2) where E is the statistical expectation operator, and = El ln]) rae = Eldtnlxtn}] Equation (2) represents a first-order difference equation in the mean value E[¥[n]]. For this difference equation to be convergent (i.e., for the system to be stable), we require that <1 |1 -n03| or equivalently @ — 1-por<1, ie, p>o (i) -1+por= i/(at1)); out (index) = (r(1) + i) * index, /index;Answer to Problem 4.38 Figure 4: Eye pattern for a=0 257wie) Figure 2: Bye pattern for a=0.05 Figure 3: Eye pattern for a=0.1Figure g : Bye pattern for a=0.2 Problem 4.39 Matlab codes % problen 4.39, cS: Haykin % oot raised-cosine and raised cosine sequences %M. Seliathurad, Datas[1 0 1100)"; 4% sample frequency 20 sample_freq=20; Ygenerate antipodal signal syms=PAM_nod(Data, sample_treq, 2); i, root raised cosine pulse rer = raisecos_sqrt(syns, sample freq ); 4% normal raised cosine pulse Fem raisecos_n(syms, sample_treq ); % plots telength(r_e_r)-1; figure; hold on 259plot (0:4/20:+/20, x Plot (0:1/20:+/20, x xlabel (’time’) Legend('root raised-cosine’ , raised-cosine’) hold off 260function osyns = raisecos_n(syms, sample. % function to generate raised-cosine si % used in Problem 4.39, CS: Haykin WM. SeLlathurai req ) 4% size of data Lisyms, w_syms] = size(syms); 4% data Reo We =[3, 343]; % Calculation of Raised cosine pulse W_T(1) = -abe(W_T(1)); time_T = [0 : 1/sanple_treq : max(W_T(2), abs(W_T(1)))]; time_TLR = R + tine_T; den = 1 - (2 * time indext = find(den’ index2 = find(den when denominator not equal to zero b(indext) = sinc(time T(indext)) .* cos(pi * time_T_R(indexi)) ./ den(index1); % when denominator equal to zero, (using L’Hopital rule) if “isempty(index2) b(index2) = 0; end; (o(sample_treq * abs(W_T(1))+1 : -1: 1), b(2 : sample.treq + ¥_T(2)+1)]; beb(:)"s % filter parameters order= floor(Length(b)/2); bbe Ol; for i= 1: order bb = [bb; b(1+i:order+i)]; end; fu, 4, v] = eva(op); d= diag(a); index = tind(d/a(1) < 0.01); if isempty(index) © = Length(ob) ; else 261ona; aa ut Length(bb)-1, 1 : 0); vi Length(bb)-1, 1: 0); u2 Jength(bb), 1: 0); dd = sqrt (4(1:0)); vdd = 1 ./ aa; uu = ut? * uz; at = uu .* (vad * dd’); a2 = dd .# via, 3)": a3 = ui(i, :) .* dd’; (num, den] = ss2tt(at, a2, a3, a4, 1); fsyms = zeros(1_syns+S+sample_freq, ¥syns) for i= 1: sampletreq : l_syns toyns(i, syme(i, end; % tittering for i = t:vsyas fsyns(:, i) = filter(num, den, feyma(:, i)); end; osyns = fsyns(( (3 - 1) © sample_treq + 2):(size(feyme, 1) - (sample_treq - 1)), :); 262function osyms = raisecos_sqrt(syns, sample_treq ) 4% function to generate root raised-cosine sequence % used in Problem 4.39, CS: Haykin 4. Sellathurai size of data Cisyms, wsyms] = size(syms); % rollott tactor R=0.3; % window WT=[3, 383]; 4% Calculation of Raised cosine pulse WT(1) = -abs(W_T(1)); timeT = [0 : 1/sample_treq : max(W_T(2), abs(W_T(4)))]; indext = tind(den den = 1 ~ (4 + tine TeR). index2 = find(den % when denominator not equal to zero b(indext)=( cos((1 + R) * pi * time_T(indext))+ (sinc((1-R)stime_T(indexi))*(1-R)*pi/4/R))./den(indert)*4#R/ pi; % when denominator equal to zero t=\pm T/4/alpha if “isempty(index2) (1+2/pi)*sin(pi/4/R)+(1-2/pi) #cos (pi/4/R))*R/eqrt(2) “RHMR/pi; Ye=0; b = (b(sample_treq * abs(W_T(1))+1 4), b(2 : sample_treq * ¥_T(2)+1)]; order= floor(1ength(b)/2); bb= tl; for i order bb = [bb; b(tti:order+i)]; end; tu, 4, v] = sva(op); a= diag(a); 263index = find(d/a(1) < 0.01); if isempty (index) © = Length(bb) ; elee 0 = index(1)-1; end; ad = bb(1); ui vi 42 F Length(bb)-1, 1 : 0); Length(bb)-1, 1: 0); lengtn(bb), 4 : 0); dd = sqrt(4(1:0)); vad 7 aa; uu = ul?» 02; at = uu .* (vdd * dd’); a2 = dd .* vi(t, :)?; a3 = ul(1, :) . dd’; (num, den] = ss2tt(ai, a2, 03, a4, 1); ‘fsyns = zoros(1_syne+3+sample_freq, w_syns); for i= 41: sample_treq : 1syms feyms(i, :) = syms(i, end; % filtering for i= t:¥syms feyms(:, i) = filter(num, den, fayme(:, i); end; osyms = feyns(( (3 - 1) + sample troq + 2):(size(tsyns, 1) - (sanple_treq - 1)), 264Answer to Problem 4.39 ro0t raised-cosine faiged-cosine Figure 1: Raised-cosine and root raised-cosine pulse for sequence {101100} 265Problem 5.1 (a) Unipolar NRZ code. ‘The pair of signals s)(2) and so(f) used to represent binary symbols 1 and 0, respectively are defined by y(t) where E}, is the transmitted signal energy per bit and 7; is the bit duration, From the definitions of s1(f) and s)(0, itis clear that, in the case of unipolar NRZ signals, there is only one basis function of unit energy. The basis function is given by OsrsT, ‘Then, we may expand the transmitted signals s;(¢) and s9(¢) in terms of 6,(t) as follows: 5() = JE,O,(9, O N. The inner product of the pair of signal s,(0) and s,(2) is given by 1 Jsosnar 6) 0 By substituting (1) in (5), we get the following result for the inner product: Tew N [Ss9,0] [Zamofe i=l oljat non of =D Ysjsufolnotnae © 0 1 ie Since the 9,(1) form an orthonormal set, then, in accordance with the two conditions of Eq. (3) and (4), the inner product of s(t) and s4(t) reduces to r Ny Jsise@ar = Ysys4j 0 jst 280T. = 8) (b) Consider next the squared Euclidean distance between s; and s;, which can be expressed as follows: IF = 6:50" (s.-s) Is- = 8/5, + 5,8, — 287 5, 7 T r Jstlanae + fsp(ende -2[5,(2)sae ° ° ° " r = f-5()"at ° Problem 5.9 Consider the pair of complex-valued signals s,() and s2(1), which are defined by S(t) = 44104(2) + ay202(1) ay S21) = a310)(1) + ay909(t) 2) The basis functions 6,(¢) and 03(¢) are real-valued and the coefficients a1, a1, a1 and ayy are complex-valued, We may denote the complex-valued coefficients as follows: x2 = O92 + B22 On this basis, we may represent the signals s,(1) and s3(t) by the following respective pair of vectors:| Oy ay = [Bul gy = {Be 2) O29] Bus Bn ‘The angle subtended between the vectors gy and gp is r 2182 cos = led Teo) To gat ap ACB + O12 + Biz + 4021 B2i + Op + Boy H, si'sy * Bl Ts a 2) where 8) = [ and sy = are complex vectors. Recognizing that 12 2. cos = I we may go on to write Jsi@synae < = 7S! re (aye | J ora iu ps00Fa) < = The complex form of the Schwarz unequality becomes J isynaa 2 = - £ J fsioPae f jsp(o/ParThe equality holds when s;(t) and s2(1) are co-linear, that is, s(t) = ksa(t) where k is any real- valued constant. Problem 5.10 EQ WCE) = ELC, + WW CELDT Els, W(t) = 9,5 EWE] = We also note that N Wt) = MCL = EW oy (ty) ist We therefore have BOC W(t, )7 = ECW, WCET N 7 - 26 ECW W(t) me apt ELW,W,1 1 T B W. Wt, = t, Ww jdt) = ELW(t, )W(t)]dt ut EL 5 IC mE ELW( 1 - Ct) ot) J 45 oo) CWC m 1 T Ny Ny 2) + 2 acetate = 38 a $36 ) (tt ” it 5st Mo, tes 2 ELW,W,1 = oO i¥j Hence, we get the final result Ny x EX 4g WCE] = r P ayty 3 20.Problem 5.11 For the noiseless case, the received signal r(1) = s(),0<1. x2. On the other hand, if x > x), it decides in favor of s2(0). If x decision is made by tossing a fair coin = xp the (b) Energy of signal s,(t) is given by r or =Jo Pare f (- 1dr + far ‘di ° r ar Energy of signal s3(0) is 1” ara sr ar = Ie irae f (uyrare f Curae f arar 12 ar st =30=E The orthornomal basis functions for the signal-space diagram of these two orthogonal signals are given by a = 22 and st) 0 = BE The signal-space diagram of signals s1 and sy is as follows: 286= 4 Br ‘The distance between the two signal points s(t) and s(t) is d= .2E = J6T The average probability of error is therefore G ae) a x a 1" oe s 4x 107 " Problem 5.13 Energy of binary symbol 1 represented by signal (0) is, 12 r = fGias f Caw = 7 ° 12 Energy of binary symbol 0 represented by signal s3(¢) is the same as shown by 287The only basis function of the signal-space diagram is s(t) _ sy) oD = TE, WT ‘The signal-space diagram of the Manchester code using the doublet pulse is as follows: Hence, the distance between the two signal points is d = 2./T’. The average probability of error over an AWGN channel is given by a ent) sere [z-) HT) Problem 5.14 (a) Let Z denote the total observation space, which is divided into two parts % and z, Whenever an observation falls in zy we say Hos and whenever an observation falls in 2, we say He Thus, expressing the risk R in terms of the conditional Probability ante functions and the decision regions, we may write Ha (ait ox Cog Po Az f, 00 Po 4z.x1H,#49 Po Sy fyyy CEH Ax 1='o + Cy, By Fo Fa, tHe + oy Py So fa, EI a For an N-dimensional observation space, the integrals in Eq. (1) are N-fold integrals. To find the Bayes test, we must choose the decision regions Z, and Z, in such a manner that the risk R will be minimized. Because we require that a decision be made, this means that we must assign each point x in the observation space Z to Z, or Z,; thus Zety+2, Hence, we may rewrite Eq. (1) as R = PgCgg Fafayy MMorAx + Poli 9 Fan fxjn, Hoe + PyC,, SF, a2g ty, 2M PE + Piloy Foye, EM ae (2) We observe that FL xyy EModee Fagin, ela = 1 Hence, Eq. (2) reduces to {P9699 Fy SIO? + TPC y gy AMD, @) The first two terms in Eq. (3) represent the fixed cost. The integral represents the cost controlled by those points x that we assign to Z). Since Cy > Coy and Co, > C,,, we find that the two terms inside the square brackets are positive. Therefore, all values of x where the first term is larger than the second should be included in Z, because the contaibute a negative amount to the integral. Similarly, all values of x where the second term is larger than the first should be excluted from Z, (i.e., assigned to 2,) because they would contribute a positive amount to the integral. Values of x where the two terms are equal have no effect on the cost and may be assigned arbitrarily. Thus the decision regions are defined by the following statement: If 289PCC, (XH) > P96, ot Fu, 10°00) fx 4(EIo) + assign x to Z, and consequently say that H, is true. If the reverse is true, assign x to 2, and say Hy is true. Alternatively, we may write fyyq CMH) x1, SMY Bt pec aby) PyCCy4-C 4) Fy lA) # > < 21Hy i 0 The quantity on the left is the likelihood ratio: Fan et? Ap Tq Zp) tH 2MHo Let accu Cre) Pip ,C 44) Thus, Bayes criterion yields a likelihood ratio test described by AGO (b) For the minimun probability of error criterion, the likelihood ratio test is described by #. inca) ea eee . H -Thus, we may view the minimum probability of error criterion as a special case of the Bayes criterion with the cost values defined as S00 = 811 =? “10 = Sor That is, the cost of a correct decision is zero, and the cost of an error of one kind is the same as the cost of an error of the other kind. 200Problem 5.15 From the signal-space diagrams derived in the solution to Problem 5.1, we immediately observe the following 1. Unipolar NRZ and unipolar RZ codes are non-minimum energy signals. 2. Polar NRZ and Manchester codes are minimum energy signals. Problem 5.16 ‘The orthonormal matrix that transforms the signal constellation shown in Fig. 5.11(a) of the textbook into the one shown in Fig. 5.11(b) is 2 1 A ° fl Sl Ble To prove this statement, we note that the constellations of Fig. 5.11(a) is defined by the four points {(ct, &), (0, ), (ot, ~01), (ct, —ct) }. The new constellation is defined by S;, rotate = QS,, which for i= 1 yields ee nw[# BIg (a (0 Similarly, 5» josie = | 5) Hence, the transformation from Fig. 5.11(a) to Fig. 5.11(b) is given by Q, except for a scaling factor. 291(a) The minimum distance between any two adjacent signal points in the constellation of Fig. PS5.17a of the textbook is d® = 20 ‘The minimum distance between any two adjacent signal points in the constellation of Fig. P5.17b of the textbook is de = A(fBay’ + (Bay which is the same as d‘“). Hence, the average probability of symbol error using the constellation of Fig. P5.17a is the same as that of Fig. P5.17b. 2a (b) The constellation of Fig. P5.17a has minimum energy, whereas that of Fig. P5.17b is of non- minimum energy. Applying the minimum energy translate to the constellation of Fig, P9.17, which involves translating it bodily to the left along the 6)-axis by the amount V/2o., we get the corresponding minimum energy configuration: Problem 5.18 Consider a set of three orthogonal signals denoted by{s;(1)}7_g, each with energy E,. The average of these three signals is a(t) Ls Applying the minimum energy translate to the signal set {sj(1)}7_y, we get a new signal set defined by 8s() = s()-a(t), 1 = 0,1,2 on) The signal energy of the new set is B= f(a =f side 2” sacar + f° ated Fn - a@ny(s()-a(n)yat 5 FELL seos nd [° ancy sfondi [a nat) Q) Since s,(¢) and s(t) are orthogonal by choice, Eq. (2) reduces to ieee eh) 1738s * 5GED) which is the maximum negative correlation that characterizes a simplex signal with M = 3. thus, 2 the signal set {sj(t)};.9 defined in Eq. (1) is indeed a simplex signal. 2932 To represent the signal set {sj(t)}ju9 in geometric terms, we use the Gram-Schmidt orthogonalization procedure. Specifically, we first set sol) G01) = a or equivalently so(t) = JEGo(0) The projection of s(t) unto go(t) is so =f si(oo(nar = Fpl ntna 5 AEE sicnsgtnar) “(4 ‘The second basis function is therefore 1) = Siobo() JE-Sio = 310) + CLE/2)(G0(0) JE = (E74) . FAloitn + Fon) Accordingly, we may express s}(t) in terms of the basis functions go(#) and 6(1) as 204 GB)s()) = ons FE on ) ‘The remaining signal s3(t) may be expressed in terms of @o(t) and (1) as 0 = Foon 6) Thus, using Eqs. (3) to (5), we may represent the simplex code by the following signal-space diagram: Problem 5.19 (a) An upper bound on the complementary error function is given by exp(-w’) anu Hence, we may bound the given P, as follows: erfe(u)< ox ( [E, Ny 1 E, N enfel [P| <= Sexo(—5t) x Je a (| 2fmE,Ny 2 PW) * ine, 7 For large positive w, we may further simplify the upper bound on the complementary error function as shown here: 2952 erfe(u) < PCH) Correspondingly, we may bound P, as follows: exp(-E,/No) pele 2) re ae Q) (b) For By/No = 9, we get the following results: (i) The exact calculation of P, yields Jerfe(3) = 10x 105 (ii) Using the bound in (1), we have the approximate value: = 116 x10 Gi) Using the looser bound of (2), we have _ xp(-9) Wn = 3.48x 10° As expected, the first bound is more accurate than the second bound for calculating P,. Problem 5.20 According to Eq, (5.91) of the textbook, the probability of error is over-bounded as follows:P= 120,M w where dy is the distance between message points s; and sy. With the M transmitted messages assumed equally likely, the average probability of symbol error is overbounded as follows: Q) ‘The second line of Eq, (2) defines the union bound on the average probability of symbol error for any set of M equally likely signals in an AWGN channel. Equation (2) is particularly useful for the special case of a signal set that has a symmetric geometry, which is of common occurrence in practice. In such a case, the conditional error probability P,(m)) is the same for all i, and so we may simplify Eq. (2) as is G <4 5 o{ “| for all i (3) f P, = P(m)) ik 2JNo, ‘The complementary error function may be upper-bounded as follows: dy) 4 di) erfe < Lexy SH (5 Ny) Jn No} Hence, we may rewrite Eq. (3) as th ora P.<—= Y, exp)-—“] for all i @) aed oom) Kal 297Provided that the transmitted signal energy is high enough compared to the noise spectral density ‘No, the exponential term with the smallest distance dj will dominate the summation in Eq. (4). Accordingly, we may approximate the bound on P, as (5) where Mmin is the number of transmitted signals that attain the minimum Euclidean distance for each m;, Equation (5) describes a simplified form of the union bound for a symmetric signal set, which is easy to calculate. 298Problem 6.1 (a) ASK with coherent reception x(t) a, Decision a — fg at device s(t) Denoting the presence of symbol 1 or symbol 0 by hypothesis H, oF Hy, respectively, we may write Hyp x(t) = s(t) + w(t)” w(t) x(t) where s(t) = A.cos(2nf.t), with A, = Y2E,7T,. Therefore, x(t) s(t) dt Sy Tf 4 > E,/2, the receiver decides in favor of symbol 1, If £ < &,/2, it decides in favor of symbol 0 The conditional probability density functions of the random variable L, whose value 299is denoted by £, are defined by fy po(t!0) = —L— expt Li SE, oD 2 GE) 1 b fey tel = expl- = Li AR WE The average probability of error is therefore, E/2 Po iakyg fuiols!O4, +P, 2 fy yCttvrae b 1° F 2 1? (2)? =e exp FDO + FS oxl- Eda £,/2 ANE, f =~ ANE FS - 2 1 “ . Lo exp(- qa at NE Vang, | E,/2 o% 1 1 B erfe(s YEA7NQ) (b) ASK with noncoherent reception meee Decision detector pan In this case, the signal s(t) is defined by s(t) = A.cos(2nfit + 6) where A, = Y2 5,77, and a O< oS 28 £,(8) = O, otherwise For the case when symbol 0 is transmitted, that is, under hypothesis Hy, we find that the random variable L, at the input of the decision device, is Rayleigh-distributed: 2 ae 2 Fi yo(#lO) = sae exp(- A) LIO ot Note For the case when symbol 1 is transmitted, that is, under hypothesis H,, we find that the 300random variable L is Rician-distributed: 252, + Aer yy ae cb. £ (81) = exp(- ) Lit Rot, Wot,/2 where Ip(28A,/Ny) is the modified Bessel function of the first kind of zero order. Before we can obtain a solution for the error performance of the receiver, we have to determine a value for the threshold. Since symbols 1 and 0 occur with equal probability, the minimum probability of error criterion yields: a exp(2ha/) VRAIN, Using this approximation, we may rewrite Eq. (1) as follows: #. Aa - AT) 1 Week exp: epjizy fe i Bi » ° 0 0 Taking the logarithm of both sides of this relation, we get 4H. 12th, 1 fe a? ci “| 0 Neglecting the second term on the right hand side of this relation, and using the fact that e azn 2 = we may write oe 1, /Er, evi fre H2 2 0 1 The threshold 3 4 5 two probability density functions, as illustrated below. is at the point corresponding to the crossover between the 300 aThe average probability of error is therefore Po Pro * Pa Por Pipe f fyb tlorae 10 LIO Vetere Ls 2 co 28 ef exp(- po-)de Vegi ,/2v2 No Not 2° 22 = [-esp(- 25)] Nol, SE ‘Ob E,1,/2v2 Poy fetes 2 2.2 a2 oa? r2yy ose Ak eb £ a exp(- 2x) ToL )ae Noty Not, /2 0 Ny VEvT sare uw a2 4 a2 Tea exp(22A./No) Og yer (= Sgt) a ob ‘oe Vivek Ny 301VEST, /2V2, 3 ia Qin Agta) 4 sere vs ele ore Fr te ce) ° eb ot The integrand in Eq. (2) is the product of Y2i/A,T, and the probability density function of a Gaussian random variable of mean A.7,/2 and variance NoT,/¥. For high values of E,/Ng» the standard deviation VNpT,/# is much less than the threshold ¥E,1,/2v2. Consequently, the area under the portion of the curve from 0 to /E,1,/2v2 is quite small, that is, Po, = 0. Then, we may approximate the average probability of error as P. Where it is assumed that symbols 0 and 1 occur with equal probability.Problem 6.2 ‘The transmitted binary PSK signal is defined by JE), OStST,, symbol 1 so) - JE), OStST,, — symbol 0 where the basis function 9(f) is defined by [Feoserro ‘The locally generated basis function in the receiver is 2 [Femerrsy [Flom eansansose = sin(2nf,t)sing] > (0) Sree(t) where @ is the phase error. The correlator output is given by Fst), 0 where x)= 5 (tw), k= 1,2 Assuming that f, is an integer multiple of 1/7',, and recognizing that sin(2nf.1) is orthogonal to cos(2nf.1) over the interval 0 <1< T,, we get _[E,cos + W ‘when the plus sign corresponds to symbol 1 and the minus sign corresponds to symbol 0, and Wis a zero-mean Gaussian variable of variance No/2. Accordingly, the average probability of error of y the binary PSK system with phase error @ is given by 303When @ = 0, this formula reduces to that for the standard PSK system equipped with perfect phase recovery. At the other extreme, when @ = 90°, P, attains its worst value of unity. 304Problem 6.3 (a) The noiseless PSK signal is given by s(t) = Acoslanf tok m(t)] = A,cos(2ntt)eoslk m(t)] - A,sin(2nf,t) sinlk,m(t)] Since m(t) = +1, it follows that Costk,m(t)] = cos(sk,) = cost k,) sinlk m(t)] = sin(+k)) = + sin(k,) m(t)sin(k,) Therefore, S(t) = A,cos(k,deos(2afgt) ~ Ant) sin(k,)sin(2ar,t) a ‘The VCO output is r(t) = Aysintant.t + 0(t)] The multiplier output is therefore 1 F(t) a(t) = 5 ALAcos(K,) {sin O(t)] + sinfane, t+0(t)]} 1 = 2 AgAyaCt) sin(k, )feost 0(t)] + cos Lal t+0(t)]} The loop filter removes the double-frequency components, producing the output 1 1 eC) = 5 AAycos(k,)sin{ 8(t)] ~ 5 A,A,m(t)sin(k, Joost O¢t)] Note that if k, = ™/2, (i.e., the carrier is fully deviated), there would be no carrier component for the PLL to track. (b) Since the error signal tends to drive the loop into lock (i.e., ®t) approaches zero), the loop filter output reduces to 1 eCt) = — 5 ALA, Sin( k, act) which 1s proportional to the desired data signal m(t). Hence, the phase-locked loop may be used to recover mt),Problem 6.4 (a) The signal-space diagram of the scheme described in this problem is two-dimensional, as shown by Quadrature | C sine) an fe This signal-space diagram differs from that of the conventional PSK signaling scheme in that it is two-dimensional, with a new signal point on the quadrature axis at A.kVT\/2. If k is reduced to zero, the above diagram reduces to the same form as that shown in Fig. 8.14. i) s(t) +w(t) Decision ay device Hy cos(29F¢) ! The signal at the decision device input is A 1, teat lia? a +4? wity costanetrat a ° e 306Therefore, following a procedure similar to that used for evaluating the average probability of error for a conventional PSK system, we find that for the system defined by Eq. (1) the average probability of error is Py = 7 erte(/e, (1-47)/Ng) 142 3 AG Ty where E, ey (€) For the case when P, = 10-* and k? = 0.1, we get 10 = 2 erfe(u) 2 0.9 W where u? = ‘0 Using the approximation 2 erfocuy = 2xp(cu) fru we obtain exp(-u?) = 2VF x 10 ° Tne solution to this equation is us 2.64. The corresponding value of E,/Ny is E D , 260 Wy = 09 Expressed in decibels, this value corresponds to 8.9 dB. (d) For a conventional PSK system, we have 1 B erte(YEL7NG) In this case, we find that 5 2 T= (2-64)* = 6.92 No Expressed in decibels, this value corresponds to 8.4 dB. Thus, the conventional PSK system requires 0.5 dB less in E,/No then the modified scheme described herein. 307Problem 6.5 (a) The QPSK wave can be expressed as s(t) m,(t) cos(2nf.t) + m(t) sin(2nf.t). Dividing the binary wave into dibits and finding m (t) and m,(t) for each dibit: dibit W 00 10 00 10 mt) E/E = VE7T Yet - VET ET mt) YEE VET ET VET - VEIT ™,(t) SE 7 bona | | 0 t Ey L| L_ mice) t ) sa) 308Problem 6.6 Let P, = average probability of symbol error in to the in-phase channel P,g = average probability of aymbol error in to the quadrature channel Since the individual outputs of the in-phase and quadrature channels are statisttcally independent, the overall average probability of correct reception is Py = (1 - Poy) A - Peg) =1- Pa - Peg + Per Peg The overall average probability of error is therefore = Par + Peg - Pet Peg 309Pre 67 Let r denote the received signal vector. Suppose that the signal corresponding to message point mis transmitted. Then, referring to the signal-space diagram of Fig. 1, the conditional probability of error is given Pp bn, = PO lies in shaded region) = P(r lies in =) + P(R lies in |) - P(r lies in ) =F erte| |= sin ® | + 1 rte 2 NM) 2 No - P(r lies in ji) Pein, < erfe ® sink Assuming that all the message points are equally likely to be transmitted, we have P, = Py|m » ™ and so Hence, 310, / / r aS Decision Z seurdany n E ne - t = Ra E r t . yl ma = 1 r| Pein 3 /| ' /! N / N / / N | : c | 311Problem 6.8 Figures 6.10 and 6.10b of the textbook, reproduced here for convenience of presentation, depict the signal-space diagrams of QPSK and offset QPSK signals, respectively: (a) QPSK (b) Offset QPSK Figure | The two parts of this figure clearly show that the signal-space structure of the offset QPSK is basically the same as that of the standard QPSK. They only differ from each other in the way in which transition takes place from one signal point to another. Accordingly, they have the same power spectral density, as shown by SUP) = Eglsine" (27 y(F-f,)) + sine? 27,(f + f.))1 where 7;, is the bit duration and f, is the carrier frequency. Problem 6.9 (a) In vestigial-sideband (VSB) modulation, there are two basis functions: + The double-bandwidth sinc function, defined by (0 = [fsine(# where T'is the symbol period and f, is the carrier frequency. cos 2/1) @ + The Hilbert transform of 6;(0), defined by92(0) = (0) = ffsine(24) sin(2nf,1) Q) where it is assumed that f, > 2/T. (Here we have made use of the Hilbert-transform pair listed as entry 1 in Table A6.4, with the low-pass signal m(t) set equal to YT7F sinc 2(t/T).) The basis functions (1) and (2) imply the use of single-sideband modulation, which (as discussed in Chapter 2) is a special form of vestigial sideband modulation. We have chosen these definitions merely to simplify the discussion. The use of VSB substitutes a realizable function for the sinc function that is unrealizable in practice. Based on the definitions of the basis functions (1) and (2) given in Eqs. (1) and (2), it may be tempting to choose 2/T as the symbol rate for successive transmission of binary symbols using binary VSB. However, such a choice of signaling destroys the orthonormality of 6,(t) and $3(0); that is, froune(s- wre 1 for j=i 0 for j#i To maintain orthogonality of (1) and 69(1), successive translations of these basis functions must be integer multiples of 1/7, as shown by f O(1)0(t-kT)dt = { nue 0 0 for j#i for any integer k. ‘Suppose, however, we restrict k to assume only odd integer values, and choose the carrier frequency f, to be an odd integer multiple of 1/27; that is, = odd integer ) We then have the following two properties 313jdt = 0 for all odd integer k 4) i) sin(2ny(r-Z)) = sin(2mft)cos(kln/2) - cos(2n/,1)sin(kin/2) = cos(2nf,t) for * = odd integer 1 = odd integer With such a choice, the implementation of the digital VSB transmission system is equivalent to a time-varying one-dimensional data transmission system, which operates at the rate of 2/T dimensions per second. (b) The optimum receiver for the digital VSB transmission system just described consists of a pair of matched filters, that are matched to the two basis functions @,(t) and 65(¢) as defined in Eqs. (1) and (2). However, in order to conform to the design choices imposed on integer & and carrier frequency f, as described in Eqs. (4) and (3), the instants of time at which the two matched filter outputs are sampled are staggered by 7/2 with respect to each other. The two sequences of samples so obtained are subsequently interleaved so as to produce a single one- dimensional data stream as the overall receiver output. The delay by 7/2 is identical to what is actually done in the offset QPSK, thereby establishing the equivalence of the digital VSB system to the offset QPSK. Problem 6.10 Assuming that modulator initially resides in a phase state of zero, we may construct the following sequence of events in response to the input sequence 01101000. Step k Phase ®.; | Input Phase Transmitted (radians) | dibit change phase 0, AO, (radians) (radians) 1 0 o1 anl4 3n/4 2 3n/4 10 -n/4 m2 3 m2 10 nia ma 4 m4 00 m4 m2 3l4Problem 6,11 The output of a n/4-shifted QPSK modulator may be expressed in terms of its in-phase and quadrature components as FE The different values of interger i correspond to the eight possible phase states in which the modulator can reside. But, unlike the 8-PSK modulator, the phase states of the 1/4-shifted QPSK modulator are divided into two QPSK groups that are shifted by 1/4 relative to each other. s(t) 10s (it/4)cos(2n ft) PEsacinraysinconr.n §=0,1,2,...7 Therefore, s(t) = FE costin/4) so(t) = PEsincineay The orthonormal-basis functions for m/4-shifted QPSK may be defined as, 91) = jPeosconren (0) = scar T ‘Then the 7/4-shifted QPSK signal is defined in terms of these two basis functions as s(t) = JEcos(in/4)o, (1) — VEsin(in/4)o,(0) On the basis of this representation, we may thus set up the following scheme for generating m/4- shifted QPSK signals: or) >| VEcos(-) 3 : Sequence of t input abit QP rites on(t) QPSK signal 3 s() Esin() 315Problem 6.12 A n/4-shifted DQPSK signal can be expressed as follows: s() = [Eoosce, "| + A0,e0s(2nf,)- PE e EE ONT where $j.1 + Ax = $4 and O,.1 is the absolute angle of symbol k-1, and Ady is the differentially encoded phase change. In the demodulation process, the change in phase $j occurring over one symbol interval needs to be determined, in (Op. + Ad,)sin(2m ft) cos (21, 14 bg, FA0,) If we demodulate the 1/4-shifted DQPSK signal using a FM discriminator, the output of the FM discriminator is given by dl2nf.t +O) You (t) = Ke [ons + 4] K(2nf, +A] where K is a constant. In a balanced FM discriminator, the DC offset 2nf-K will not appear at the output. Hence, the output of the FM discriminator is KAdy. Problem 6.13 The output of a m/4-shifted DQPSK modulator may be expressed as s(t) = Ee. +40,)cos(any.0)- PE = [Fiecso, 1CosAG, ~ sinO,., sin AO, Joos (2m ft) in(O, 1 + AO,)sin(2nf,t) in, cosAO, + 0050, SinAQ,]sin(2mf,,1) 316Let I, = cos(0,.4 +A0,) and Q, = sin(0,., +40,). We may then write 1, = cost, ,cos A, — sind, ,cosA0, = c08(0,.> +40,,;)cosA0,—sin(0,.9 +8, ,)cosAG, from which we readily deduce the recursion I, = k 4.160800, ~ 0, sin, Similarly, we may show that Q, = sin®,.cosA0, + cosO.; sinA0, = Qj.100sA0, +1, ,sinAO, From the definition of J; and Q,, we immediately see that J, and Oy may also be expressed as 1, = cosO, and which are the desired resultsProblem 6.14 339-36 91 -27 +79 19-15 -1 31 5 9 13 7 21 5 29 33 37 236 234 234 298 18 173 226 199 165 146 133, Cress Const ML Squove const QC. DT RuaiGr - supereonst lan. ¢ V.34 modem 318Problem 6.15, ‘The transmission bandwidth of 256-QAM signal is 7 2R, © Tog where Rj is the bit rate given by 1/7, and M 2(1/T p) 2 Tog,256 ~ T6T, op 256 aT, The transmission bandwidth of 64-QAM is e220) eee ot Tog364 «BT, AT, Hence, the bandwidth advantage of 256-QAM over 64-QAM is 170E, where Ep is the energy of the signal with the lowest amplitude. For the 64-QAM signal, we have _ 63 6 = 9 = 42Ey Therefore, the increase in average signal energy resulting from the use of 256-QAM over 64- QAM, expressed in dBs, is 1708, lolog (=) ih BB y) * 100804) = 6dBProblem 6,16 The probability of symbol error for 16-QAM is given by (1 we Farin) Setting P, = 10°, we get (1 jefe) Solving this equation for Eyy/No, 10 The probability of symbol error for 16-PSK is given by ext [Fsintn/a0) Setting P, 3 ( [E 10° = ert [FE sinen1s) Solving this equation for E/No, we get = 10°, we get E = = 142 = 21.5dB No Hence, on the average, the 16-PSK demands 21.5 - 17.6 = 3.9 dB more symbol energy than the 16-QAM for P, = 10°, Thus the 16-QAM requires about 4 dB less in signal energy than the 16-PSK for a fixed Ng and P, = 10°, However, for this advantage of the 16-QAM over the 16-PSK to be realized, the channel ‘must be linear. 320Problem 6.17 (@) An M-ary QAM signal is defined by 7G) cos (2m ft) - sin (2m ft) ay We can redefine the M-ary QAM signal in terms of a general pulse-shaping function g(t) as s(t) = ayg(t—kT)cos(2nf,.1)-bye(t- kT)sin(2nf,1) @) The M-ary QAM signal s(t) for an infinite succession of input symbols can be expressed as st) = DY 5,(0) DY {a,g(t-kT)cos(2nf 1) - byg(t— kT) sin(2nf,t)} knw Rel y lars sogee in| me = Re xy aucune ™| 8) where A, is a complex number defined by Ay =a, + jby By multiplying Eq. (3) by exp(j2nf,k7) x exp(i2n/,kT), we get si) = af z gtk P2afekTso( anf} - rel y Agee] ) = 321where Ay = Ayexp(j2mfkT) R(t) = g(texp(j2nf,t) The scalar A, is a rotated version of the complex representation of the Ath transmitted signal. Equation (4), representing a QAM signal, appears to be carrierless, therefore, it is equivalent toa CAP system. (b) A CAP signal is defined as s(t) = Rel SY Ager er Kew : Rel = Rel > ae—mnewtinnf9| DY Apexpi2mf AT) g(t kT )exp(j2m f(t - any} = DY ag(t~kP)cos(2nf,1) - DY, byglt-kT)sin(2n ft) () kaw Now the pulse shaping functions of CAP signal, g(1-- kT), may be replaced by and the formulaton in Eq. (5) can be rewritten as = > PRacoscansay— PE o,sincans.0 © = The Ath signal of the signal s(t) defined in Eq. (6) is given by sy(t) = 0s 2nf.1)~ PED, sin 2nf.0), Oxter @ ‘The signal formulation given in Eq. (7) is recognized as the M-ary QAM signal of Eq. (1).Problem 6.18 ‘The CAP signal can be expressed as s(1) = Y a,p(t-nT)- Db, p(t-nT) where p(t) is the Hilbert transform of the pulse p(t), and A,, = a, +0, The CAP signal s(2) can be written in the equivalent form 0-10) * pw {x 2,80-n0) p(t) where (2) is the delta function, and the star denotes convolution in the time domain. Hence, the power spectral density of s(1) is 2 2 sf) = Benes SI Fier where 62 and of are the variances of symbol a; and by, respectively, where p(t) = P(f) and B(t)# P(f). Noting that|P(/)| = P(/), we thus have 2 ot 3 pane SN) = Next, noting that 2,2 2 _1 Oo, +5, = 6, = we finally get SAA) = ORIPA?Problem 6.19 From the defining equations (6.74) and (6.75) of the textbook, we have P(t) = g(s)cos(2nf,t) and BU) = gsincansn Applying the Fourier transform to Eqs. (1) and (2), we get PU) = IG) #6F +f] and PUD = HIGU-f)-GU +f] Accordingly, we may determine p(t) and p(t) by proceeding as follows: * Given G(f), use Eqs. (3) and (4) to evaluate P(f) and P(f). * Using the inverse Fourier transform, compute p(t) = F'[P()] and p(s) = FLP(f)] 324 a) 3) (4)Problem 6.20 For binary FSK, the two signal vectors are ae ° 0 Fo | where E, is the signal energy per bit. The inner products of these two signal vectors with the observation vector 1 oc are as follows, respectively, (x58,) = x am e e "YE where (x, 8;) = x"; for i=1,2. The condition xs, >x7s. is therefore equivalent to VB, x1 > VB, x2 Cancelling the common factor VB, we get x > Xp which is the desired condition for making a decision in favor of symbol 1. 325Problem 6,21 The bit duration is Sa = OH us! 2.5x10°Hz -The signal energy per bit is x 0.4 x 10% = 2x 10'9 joules (a) Coherent Binary FSK The average probability of error is er fo(VE,72N5) ni- erto(Yaxt0~'9/ux1 x er fo( 45) ni Using the approximation 2. erfo(uy = £xp(cu") Yuu we obtain the result 1 exp(-5) -3 al = 0.85 x 10 er 2g erfe(/E7ig) = erfe(/T0)exp(-10) Vion = 0.81 x 1079 (c) Noncoherent Binary FSK 1 Po = 5 expl= 2 2Ny 1 =F exp(-5) = 3.37 x 1073 Problem 6.22 (a) The correlation coefficient of the signals sq(t) and s,(t) is 1, £? sicers(erat ° 0 1 T, eee ts? sicenata us? s®ctyata 0 O 72 T, A2 5 costan(t.s Latyt] costan(t- dart] at 2 yi2 (deg ye Oy apn)? tp net) Leos(amirt) + cos(unf,t)] dt sin(2mafT,) sin(4nfT,) 1 [ cuoee seme os omer 2 uae oO Since f, >> Af, then we may ignore the second term in Eq. (1), obtaining sin(2nAfT, ) ait ar = sine(2sen,) (>) The dependence of p on Af is as shown in Fie. 1. 327Coppelation coefficient e Fig. 1 3288,(t) and s,(t) are othogonal when p = Therefore, the minimum value of Af for which they are orthogonal, is 1/27, (ec) The average probability of error is given by er fot VE, (T=) 72Ng) nie Es The most negative value of p is -0.216, occuring at Af = 0.7/1. The minimum value of P, is therefore 1 Paymin = 3 er fe(v0.608E,7N,) (d) For a coherent binary PSK system, the average probability of error is 1 Py = 3 erfe(YEL7ND) Therefore, the £,/N) of this coherent binary FSK system must be increased by the factor 1/0,608 = 1.65 (or 2.16 dB) so as to realize the same average probability of error as a coherent binary PSK system, Problem 6,23 (a) Since the two oscillators used to represent symbols 1 and 0 are independent, we may view the resulting binary FSK wave as the sum of two on-off keying (00K) signals. One 00K signal operates with the oscillator of frequency f,- The second 00K signal operates with the oscillator of frequency f+ The power spectral density of a random binary wave X,(t), in which symbol 1 is represented by A volts and symbol 0 by zero volts, is given by (see Problem 4/0) sinc?(17,) where Ty is the bit duration, When this binary wave is multiplied by a sinusoidal wave of unit amplitude and frequency a + 4£/2, we get the first OOK signal with as EST, The power spectral density of this 00K signal equals 1 af, af, 8) 27 ts, ¢F oho + Sy (Pe te 329The power spectral density of the random binary wave Xp(t) = X,(EJ, in which symbol 1 is represented by zero volts and symbol 0 by A volts, is given by Sy (f) = Sy (f) %, x. When X,(t) is multiplied by the second sinusoidal wave of unit amplitude and frequency f, - 4£/2, we get the second 00K signal whose power spectral density equals 1 oe, Ae, S30) = GIS, (f= f+ 5) + 5, +t,- 351 The power spectral density of the FSK signal equals: Spgx(f) = $,(2) + (2) E, b ae, Ag, af, = Br, (HE fo BY + Mes tee + Or - t+ ye Ore as clea 2, af. 2, af, {sine [T,(f - £,- 51 + sine“[t(¢ +f, + 5491 2 Af, 2, + sine (T,(f = £, + 5°)1 + sine [T(f + £ This result shows that the power spectrum of this binary FSK wave contains delta functions at f= £. + Af/2. (b) At high values of x, the function sinc(x) falls off as 1/x. Hence, at high 2 frequencies, S.., falls off as 1/f°. 330Singnge) phase Shifb en Problem 6.25 ‘The similarities between offset QPSK and MSK are that both have a half-symobl delay between the in-phase and quadrature components of each data symbol, and both have the sane probability of error, The differences between the two techniques are: (1) the basis functions for offset QPSK are sinusoids multiplied by a rectangle function, while the basis functions for MSK are Sinusoids multiplied by half a cosine pulse, and (2) offset QPSK is a form of phase modulation while MSK is a form of frequency modulation. 331Problem 6.26 For coherent MSK, the probability of error is P, = erfe(vES7Ng) » while for noncoherent MSK, (i.e., noncoherent binary FSK) 2) 2H) * 1 Pg = b expl- 5 Ey To maintain Pe = 10 for coherent MSK, 79-8. To maintain the same probability of 0 symbol error for noncoherent MSK, 21.6, which is an increase of 3.4 dB. (b) te) 332Problem 6.28, (a) The Fourier transform of h(#) is given by (using entry 5 of the Fourier-transform pairs Table 6.3) ae L | Hp) = . eq Fel Te) = exp-(fra") oO Substituting a = (,/log272/W) into (1), we get Hf) = exo? 1982 1) 2) Let fp denote the 3-dB cut-off frequency of the GMSK signal. Then, by definition, |H(fo)] = 11(0)) 2 1 “OB Hence, from Eq, (2) it follows that EG) or 2 exo(ine2(29)) = 2 Taking the logarithm of both sides, we readily find that foo W 333The 3-dB bandwidth (cut-off frequency) of the filter used to shape GMSK signals is therefore W. (b) The response of the filter to a rectangular pulse of unit amplitude and duration T'centered on the origin is given by 12 ai) =f he-nae 12 3) Let k = 4) Equation (4) is finally rewritten as % ° 1) 2 2 ne 2 2 54 f exp dk + f exp(-e?)ak aA) a a) = 1 1 = ~ zerf(ky) + 5erf (ky) 3345 HL ~erfe(ty)] + su ~erfe(k,)] S Jerfo(ks) = Jerferk,) 1 a(t 7/2, Verte BE=7/2y Hone Problem 6,29 1 ost 0 ~05 a] 1 0 1 2 3 4 5 1 Imaginary part 05 ° 05 1 2 3 4 5 1 Real part 05 ° 05 a . 1 ° 1 2 3 4 5 GMSK signal 335The GMSK signal, displayed in the bottom waveform, is very similar to that of the MSK signal in Fig. 6.30, both of which are produced by the input sequence 1101000. This objective is indeed the idea behind the GMSK signal. Problem 6.30 Comparing the standard MSK and Gaussian-filtered GMS signals, we note the following: ( larities + Fora given input sequence, the waveforms produced by the MSK and GMSK modulators are very similar, as illustrated by comparing the GMSK signal displayed in the solution to Problem 6.29 and the corresponding MSK signal displayed in Fig. 6.30 of the textbook for the input sequence 1101000. + They both have a constant envelope. (b) Differences The use of GMSK results in a slight degradation in performance compared to the standard MSK for a time-bandwidth product W7,, = 0.3. However, the GMSK makes up for this loss in performance by providing a more compact power-spectral characteristic, 336Problem 6.31 In the binary FSK case, the transmitted signal is defined by cos(2rfit), O lp, the receiver decides in favor of symbol 1, and if < ly it decides in favor of symbol 0. Suppose symbol 1 or frequency f, is transmitted. Then a correct decision will be made by the receiver if l; > ly. If, however, the noise is such that 1, < ly, the receiver decides in favor of symbol 0, and an erroneous decision will have been made. To calculate the probability of error, we must have the probability density functions of the random variables Ly and Ly whose sample values are denoted by 1, and lp, respectively. When frequency f, is transmitted, and there is no synchronism between the receiver and transmitter, the received signal x(t) is of the form xit) = |B costanfyt + 6) + wit) Te (2) 2; Bb 65 0 cos(2ntyt) - | Eb TF; sin @ sin(2nf,t) + w(t), OstsT, > Let 337Gh xo 2 cos(2nft)dt, i= 12 @) b Fier matahea Sapte atime by seer Fe ty oh ae Comparison] “hoes! ss thet, chao Filter matched to a Fi 7 tOti 5) colors To osts ty mer = Ty Figure 1 and Ty 2 5 i xan fo | 2 sintntinat, t= 12, Qo The x, and x,; i=1,2, define the coordinates of the received signal point. Note that, although each transmitted signal s,(t), i=1,2, is represented by a point in a two-dimensional space, the presence of the unknown phase @ makes it necessary to use four orthonormal basis functions in order to resolve the received signal x(t). With the received signal x(t) having the form shown in Eq. (1), we find that the output of the upper channel in the receiver of Fig. 1 equals (6) 338where Xe1 = YBp cos © + Wey and Xe1 = - ¥Bp sin @ + Wer On the other hand, the corresponding value of the lower channel output is where eg = Wea and X52 = Wz The wa and w,i, i=1,2, are related to the noise w(t) as follows: = [7 wit) = cos(2afit)dt, b Wei and fh wo A sin(2nft)dt, bd (6) @D (8) (9) (10) qa) (a2) Accordingly, w,; and w,i, i=1,2, are sample values of independent Gaussian random variables of zero mean and variance Ny/2. When symbol 1 or frequency f, is transmitted, we see from Eqs. (9) and (10) that xe and x49 are sample values of two Gaussian and statistically independent random variables, X,. and X,p with zero mean and variance N,/2. Accordingly, the lower channel output 1, related to xX, and x, by Eq. (8), is the sample value of a Rayleigh-distributed random variable Lo. We may thus express the conditional probability density function of L,, given that symbol 1 was transmitted, as follows: 339fala |) = Ft exp 2 | 1,20 (a3) Again under the condition that symbol 1 or frequency f, is transmitted, we see from Eqs. (6) and (7) that x, and x,, are sample values of two Gaussian and statistically independent random variables, X,, and X,,, with mean values equal to /E, cos @ and /Ey sin 0, respectively, and variance No/2. Therefore, the joint probability density function of X, and X,1 given that symbol 1 was transmitted and that the random phase @ = @, may be expressed as follows 4) Frcartr lls Obsea Xa1 18) = Eexp]- [ceer=(/p 0080)? + (xy, + yy sin oy? |! ™No ‘0 Define the transformations Xer = 1 cos Wi y15) and Xe =] sin Vy ae) where yj = tan"(x,)/x,1), with 0 < yj, < 2n. Then, applying this transformation and following a procedure similar to that described in Section 5.12, we find that the upper channel output 1, is the sample value of a Rician-distributed random variable L,. Hence, the conditional probability density function of L,, given that symbol 1 was transmitted and that the random phase @ = 8, is given by the Rician distribution firth 1, = [* favhedn v IL, ody (a7) where 1,(21,VE,/No) is the modified Bessel function of the first kind of zero order. Since Eq. (17) does not depend on 8, which is to be expected, it follows that the conditional probability density function of Ly, given that symbol 1 was transmitted, is 3402 1+ By nS? al al, /B; 1 fyi n; 11) = 2 exp| - 1 VB | 4,20 er) No fo Note that by putting Ey = 0 and recognizing that I,(0) = 1, Eq. (18) reduces to a Rayleigh distribution. When symbol 1 is transmitted, the receiver makes an error whenever the envelope sample ly obtained from the lower channel (due to noise alone) exceeds the envelope sample 1, obtained from the upper channel (due to signal plus noise), for all possible values of 1,. Consequently, the probability of this error is obtained by integrating f; | ,(1, | 1) with respect to ly from 1, to infinity, and then averaging over all possible values of l,. That is to say, Po; = P(l>l; | symbol 1 was sent) (19) =f aiiin (deft, nde ID where the inner integral is the conditional probability of error for a fixed value of 1,, given that symbol 1 was transmitted, and the outer integral is the average of this conditional probability for all possible values of 1,. Since the random variable L, is Rayleigh-distributed when symbol 1 is transmitted, the inner integral in Eq. (19) is equal to expl,/24y. Thus, using Eq. (18) in (19), we get . ‘eo 21; 2; + Ey 21, yE (20) rat Real a Define a new variable v related to 1, by (21) Then, changing the variable of integration from 1, to v, we may rewrite Eq. (20) in the form ee 24 a2 (22) Pa don m)6 von) 8" ade where a = VE,/No. The integral in Eq. (22) represents the total area under the normalized form of the Rician distribution. Since this integral must be equal to one, we may simplify Eq, (22) as 341(23) Similarly, when symbol 0 or frequency fy is transmitted, we may show that pig, the probability that 1 > ly and therefore the probability that the receiver makes an error by deciding in favor of symbol 1, has the same value as in Eq. (23). Thus, averaging pio and po), we find that the average probability of symbol error for the noncoherent binary FSK equals (24) which is exactly the same as that in Eq. (6.163) in the textbook. Comparing the effort involved in the derivation of Eq. (24) presented in this problem with that in deriving Eq. (6.163), we clearly see the elegance of the approach adopted in the textbook. 342Problem 6.32 Let x(t) A,cos(2nf.t + 8) A,cos(2nf.t) cos 6- A,sin(2nf.t) sin 6 The output of the square-law envelope detector in Fig. P@*2, sampled at time t=T, is given by r op 2 Ly x(t) cos(anf ty dtl + (F x(t) sin(2nf.t) dtl o ° o ° y(T) This may be written as Th YT) = SS x(ty x(t, )Loos(2af,t, oos(2af ty) + sin(2af.t,)sin(2af tp) ]dt dt, 00 - Put t, = t, and ty = tex. This transformation is illustrated below: Then, we may rewrite Eq. (1) as follows T Tt y(T) = Sf x(t) x(ter) Loos(2mr teos(2mf t + 2mf,1) nee e e e + sin(2uf.t)sin(2af.t + 2xfjx)] dt dr However , cos(2nt Hoos(2nt.t + 2xf x) + sin(2nft)sin(2at.t + 2af 1) = cos(2af,D Therefore, we may simplify Eq. (2) as follows T TH y¥(T) = a J, x0t) x(tor) cost2at,n) dr dt T Tt ~ oS = x(t) x(tet) cos(2af,t) dt dt, OC r 2. However, for large values of E/Ng and M > 4, the probability of symbol error is approximately given by a) For coherent M-ary PSK, the corresponding formula for the average probability of symbol error is approximately given by @ (a) Comparing the approximate formulas of Eqs. (1) and (2), we see that for M > 4 an M-ary DPSK system attains the same probability of symbol error as the corresponding coherent M-ary PSK system provided that the transmitted energy per symbol is increased by the following factor: (b) For example, (4) = 1.7. That is, differential QPSK (which is noncoherent) is approximately 2.3 dB poorer in performance than coherent QPSK. 347Problem 6.35 (a) For coherent binary PSK, =i : =10 For P, to equal 10°", YE,7W, = 2.64. This yields E,/Ny = 7.0. Hence & = 3.5 x 107°, ‘The required average carrier power is 0.35 nil. (b) For DPSK, 5 1 3 exp(- _) - 2 Ny 5, For P, to equal 10°, we have q2= 8.5. Hence £, = 4.3 x 107°, me required average 0 power is 0.43 mW. Problem (a) For a coherent PSK system, the average probability of error is 1 nee Pay erfol VEN] 4 xPC~CE,/Ny) 4] +} —= VED), a) For a DPSK system, we have P, 1 e =D exPL= (EL/Ny)9] (2) Let 3485 g, GB = GE) +6 M2 4 Then, we may use Eqs. (1) and (2) to obtain Vi ETM), = exp 5 We are given that an YT.2R) = 1.56 ‘Therefore, 5 10 log, (<2) = 10 log, 7.2 = 8.57 4B 10%), 10 et 10 108,017 10'F ‘2 10 10g, 9(7.2 + 1.56) 9.42 eB The separation between the two (E,/N)) ratios is therefore 9.42 - 8.57 = 0.85 dB. (>) For a coherent PSK system, we have B ertel VEN); ] 4 OxPL=(E,/Ny) 4] “2 Se ae (E/N), For a QPSK system, we have 3) er fet YTE MN ¥) expl-(E,/Ny) a] A VETN2 Here again, let a) E Fe Then we may use Eqs. (3) and (4) to obtain 349exp(-5) We 7N), T+ SEND), Taking logarithms of both sides: z (5) win 22 - = 0.5 £nlt + 5/(E,/No) 4] = 05 —— EAD, Solving for 6: ae an 2 T+ 0.5/CE,/No); = 0.65 Therefore, 10 10g49(7+2) = 8.57 aB 10 10g, 9(7-2 + .65) 8.95 4B. The separation between the two (E,/N,) ratios is 8.95 - 8.57 = 0.38 dB. (c) For a coherent binary FSK system, we have Py = $ erfelVtE,7aR),} 1 exp(- 4%) ) 2 Noy 73S (6) Va YES7ENG), For a noncoherent binary FSK- system, we have 1 b> B exrl- 3G) ) mM 2 2 No 2 (8) We are given that (E,/N)), = 13.5. Therefore, 3506 = en( 33) = 3.055 We thus find that 10 108, oe) 10 og, (13.5) 1D 10 11.3 dB 10 reese), = 10 1og,9(13.5 + 3.055) 12.2 dB “Hence, the separation between the two (E,/N)) ratios is 12.2.= 11.3 = 0.9 4B. (a) For a coherent binary FSK system, we have 1 5 erfel VE, 72NG); E 1 exp(- 32) ) 1 2 Mon 2 RETR), (9) For a MSK system, we have re Py = 3 erfel VCE 72No)2] (0) = 02 (10) Hence, using Eqs. (9) and (10), we 6 1 aD) +3 an 2-d anti + Gg EN) a) Noting that 6 ea“! aD) we may approximate Eq. (11) to obtain 1,8 i in2-$6 1=36 ay EARL Pe Solving for 6, we obtain 351= 1.29 We thus find that 10 log, (<8) = 10 log, (13.5) = 10 x 1.13 = 11.3 4B 10K), 10 E 10 log, 4(g2)_ = 10 log, 9(13.5 + 1.29) = 11.7 dB 10'%y) > 10 Therefore, the separation between the two (E,/Nq) ratios is 11.7 - 11.3 = 0.4 dB. 352Problem 6, os Noncoherent FSK (a) Coherent PSK’ Coherent SK (6) Coherent MSK. | | | | Coherent QPSK oPsk 10° | 1 I | a TS bo Figure 1 Comparison of the noise performances of different PSK and FSK systems. The important point to note here, in comparison to the results plotted in Fig. _is that the error performance of the coherent QPSK is slightly degraded with respect to that of coherent PSK and coherent MSK. Otherwise, the observations made in Section 8.18 still hold here. 353Problem 6 The average power for any modulation scheme is This can be demonstrated for the three types given by integrating their power spectral densities from -« to =, Pes S(f) af a FTL (p(t - £) + sgt + toler oF 2 / Splfar The baseband power spectral densities for each of the modulation techniques are: | PSK apsk sk 2 2 32E,, (cost2xfT,)| Set) | 26, sinc(r7,) 4B, sino carn,» = —s err? 4 L E Since [" a sind(ax)dx = 1, P 7, is easily derived for PSK and QPSK. For MSK we have a > 354r 2 . ? 168, eos(2nfT,,) | Ps fs \ at we | 16f7r2 - 1] cos@(2mx) : Tee ae (16x = 1) By 7 1+ costume) 4, * er 2 (2 1 * wT, -* 16x" (x> = %e Ey cos 0 (ax) . 7 908.0 + costum) gy eetentae 12 WewT, 2 OP ~ ap From integral tables, (see Appendix Ail-6) x £ S98(8)4F | a tatn(ab) - abcos(ad)] ow x)? OY For a= 0, the integral is 0. For a= 4x, b= qy we have Eee ae E b cos(ax) b P= f SORA) gy 2D 163" 2) T, Hence, the noise equivalent bandwidth for each technique is as follows: PSK oPsk SK ; j 1 1 Bandwidth | st % aT 355Problem 6.39 (a) Table 1, presented below, describes the differential quadrant coding for the V.32 modem of Fig. 6.48a in the textbook, which may operate with nonredundant coding at 9,600 b/s. The entries in the table correspond to the following: Present inputs: Q4,0> Previous outputs: 1.1 Present outputs: ty, Table 1 Input Previous output Present output dibit dibit dibit Qin Qn | lint | lant Tin lay 0 1 0 0 0 0 0 L 0 1 0 7 0 1 1 0 1 0 0 I 7 - 1 1 0) 0 0 0 0 1 0 oO 0 1 1 1 0 0 1 0 oO 0 0 0 1 1 1 0 1 0 0 0 1 1 1 0 0 1 1 0 1 0 1 0 0 1 1 0 1 I 0 oO 1 1 0 0 1 0 1 1 0 1 0 0 1 1 1 0 1 1 1 1 1 1 0 1 356(b) Table 2, presented below, describes the mapping from the four bits I) ».172 p-1+ 03,04,» t0 the PI tnetl2n-t» 0312s, output coordinates of the V.32 modem. Table 2 Present output | Present input Output dibit dibit coordinates Tin lon Q3n 24. o 2 0 1 0 0 1 -l 0 1 0 I 1 a 0 = 1 0 3 -1 0 BE 7 I a 3 0 0 0 0 “ll “1 0 0 0 1 3 Bt 0 0 1 0 1 3 0 0 1 1 3 3 1 0 0 0 <1 1 1 0 0 1 <1 3 1 cv) 1 0 3 1 1 0 1 1 3 3 1 1 0 0 1 1 1 1 0 1 3 1 1 1 1 0 1 3 1 1 1 1 3 3 (b) We are given the current input quadbit: 21, nQ2,nQ3,nQs,n = 0001 and the previous output dibit Tin-tlan From Table 1, we find that the resulting present output dibit is Tinton = 4th quadrant 3rd quadrant } / 2nd quadrant Ist quadrant Hence, using this result, together with the given input dibit Q3 ,04, = 01 in Table 2, we find that the coordinates of the modem output are as follows: 357o, = 3, and = 1 We may check this result by consulting Table 6.10 and Fig. 6.49 of the textbook With Qj, ,Q2y = 00 we find from Table 6.10 that the modem experiences a phase change of 90°. With Fry Foyer = Ol, we find from Fig. 6.49 that the modem was previously residing in the fourth quadrant. Hence, with a rotation of 90° in the counterclockwise direction, the modem moves into the first quadrant. With Qs ,Qs,, = O1, we readily find from Fig. 6.49 that 6, =3, and = 1 which is exactly the same as the result deduced from Tables 1 and 2 of the solutions manual. For another example, suppose we are given Qj, nr, nQ3,nOa,n = LOU and Ty nal ‘Then, from Table 1, we find that Fnlajn = 00 Next, from Table 2, we find that the output coordinates are $y =-3 and 2 = -3. Confirmation that these results are in perfect accord with the calculations based on Table 6.10 and Figure 6.49 is left as an exercise for the reader. Problem 6.40 (a) The average signal-to-noise ratio is defined by Pay (SNR) gy = @ ° where Pyy is the average transmitted power, and 62 is the channel noise variance. The transmitted signal is defined by sylt) = a,cos(2nft)—Bysin(Q2n fF), OStSTwhere (ap, bs) is the kth symbol of the QAM signal, and T is the symbol duration. The power spectrum of s;(t) has the following graphical form: Power spectrum of 540) Main lobe Fig. | On the basis of this diagram, we may use the null-to-null bandwidth of the power spectrum in Fig. 1 as the channel bandwidth: Py = 2Ey = —™ @ where E,, is the average signal energy per symbol. To calculate the noise variance 6, refi» the following figure: Pox spectrum of noise | No/2 0 ie 5 Lo .s_ Fig. 2 359‘The noise variance is therefore o = NB GB) Hence, substituting Eqs. (2) and (3) into (1) BE,,/2 (SNR)qy = 5 NB 1/E aw * alm) Expressing the SNR in decibels, we may thus write Ew 10log (SNR), = —3 + 10log (se).08 10\No. Given the value 1010819, (Eay/No ) = 20 dB or E,y/Np = 100, we thus have 10log jg(SNR),y = 17 4B (b) With M= 16, the average probability of symbol error is, P,= 1 we farim| 1 100 E ot serte( Hm) = 1.16% 10° Problem 6.41 We are given the following set of passband basis functions: i {O(t)eos(2nf,1), $(4)sin(27f,1)}, 360Property 1 fi (o(Heos(2nf,M)PU(e)sin(2mf,N))dt = 0 for all n a To prove this property, we use the following relation from Fourier transform theory: feos ans, O(sin(2xf,)de = 0 for all n @ where g,(t)G,(f) for i = 1,2, and the asterisk denotes complex conjugation. For the problem at hand, we have att) = [Psine((A)cos(2nF,0) a(t) = Fsine((2)sineens,0) ‘The Fourier transform of the sine function is F [sine{2)] = Trect( fT) where 1 1 hspsh rect(fr) = | | for -aRSfS5p 0 otherwise rect((f—f,)T) + rect((f + f,)T)] 361rect((f-f,)T) — rect((f + f,)T)] Let 7, denote the integral on the left-hand side of Eq. (1). We may then use Eq. (2) to write AG) tech -F,7) ~ reel + far ® where the integrand is depicted as follows: 4 : f Fig. 1 fy ° : — VT le— VT —s From Fig. | we immediately see that the areas under the two rectangular functions are exactly equal. Hence, Eq, (3) is zero, thereby proving Property 1 for any n. Property 2 fi Jaf! 1 PRE 1 for k= n ole on dt = 4) Late (Z oe) { 2 for ken ‘ Let J denote the integral in Eq, (4). When k = n, we have ~ Loop = af ear = slo (9) 3621 of = Zf_sine (Za Using Rayleigh’s energy theorem, namely, fear = we may write 6nPar 3A sine?[Z)at = sine" an, hewT fi reo (nar which proves Property 2 for k = To prove Property 2 for k#n, let par, gilt) = one” _P Ward : FFsne(S)e Dap a2) = OMe fesine($) ttle Then applying the following relation from Fourier transform theory, Feanewo a =f ene af we may rewrite the integral J, of Eq. (4) asar F[sine (pe) F[sine (2) eo ap Since f,, = n/T by definition, we may depict the two Fourier transforms constituting the integrand of Jy as shown in Fig. 2 for the worst possible case of k= n+1: Flsince ye nly Fig.2 Hf gt ft 2 2s From Fig. 2 we immediately see that the Fourier transforms of (1) and 9(t)e will never overlap for k #. Hence, the integral J; is zero, proving the rest of Property 2. Property 3 Forney * oayen(nye yah =0 for ken where the star denotes convolution. From the convolution theorem, we have FLO@)*A()] = O(AHCS) where ®(f) = F[(t)] and ®(f) = F[h(1)]. For k #7, the picture portrayed in Fig. 2 remains equally valid except for the fact that the basic rectangular spectrum is now replaced by that rectangular spectrum multiplied by the frequency response H(f). This multiplication does not affect’ the nonoverlapping nature of -—sthe_~— two spectra._—_representing Coens ana (QUp*ACNE” *) for k #n, hence, proving Property 3. Problem 6.42 Step 1 - Set k = 0 and the initial noise-to-signal ratio NSR(K) = 0. Sort the subchannels used in ascending order (i.., from the smallest to largest ones), Step 2 - Update the number of subchannels used by setting k= k+1. 364Step 3 - Compute NSR(k+1) = NSR(O) + 8 Step 4 - Set 2(k) = PP ESRD) % Step 5-If Py = A(k)-T] | <0 Bk. (st) then compute P; = A(k-1) A 5 Pi) and for 1 = 1,2,-,k-1 Otherwise, go to step 2. For notations, refer to Section 6.12. (The algorithm presented here is adapted from T. Starr, .M. Cioffi, and PJ. Silverman (1999); see the bibliography.) Pr 43 (a) P| +P, +P3 a o 2 | -K = TS =To* @ 81 P,-K = ) 365@) ‘Adding Eqs. (2), (3) and (4), and then using Eq, (I): 2 lot 3K = P40 ee 40° (1+ i +7) Solving for K, we thus have 10,1 yt 3(1 +3-3) a 3 366Problem 6.44 (a) Using matrix notation, the channel output vector is defined by x= Hatw where H is the channel matrix, a is the transmitted signal vector, and w is the channel noise vector. Applying the singular value decomposition to H, we may write a = ufa‘o]vi where + denotes Hermitian transposition. The vector-coding receiver uses a bank of discrete matched filters defined as the rows of orthonormal matrix U*. Thus passing x through this bank of matched filters, we get X = Ux = Ul(Ha+w) = ui(ufa olvia+w) a Defining A=Via and recognizing that UU = I (identity matrix), Eq. (1) reduces to X= AA+W Q) where 367W = Ulw Each element of the vector X in Eq. (2) represents an independent channel, and so we write p= AyAyt Wy = 1,2, (b) In the multichannel transmission model, the channel capacity of the entire system in bits per transmission is given by Ins-Z3)) (3) inst where v is the length of the channel impulse response. We may also express the R as follows: R = Flog (1+ ESNR) coe cating) ° Hence, combining (3) and (4): Ne p \\OR P+ (SNR) pce enee = FTI 1+ 24 | dnt Toy (SNR) vector coding if (©) As the block length goes to infinity, we may ignore v, in which case the channel matrix H becomes nearly an N x N matrix. Therefore, H may be decomposed as H = Q"aQ 368where Q is an orthonormal matrix, and A is a diagonal matrix of eigenvalues (ie., singular values). Correspondingly, the singular values approach the magnitude of the Fourier transform, Even though a vector coding receiver and discrete multitone receiver converge to the same performance, they are not the same: +The subchannel gains are the same in both cases, but in vector modulation all subchannels have zero phase, while in DMT the subchannels have arbitrary phase angles + Unlike DMT, a vector-coding system does not require the use of a cyclic prefix. +The computational complexity of the vector-coding multichannel system is much greater than that of its DMT counterpart. Problem 6.45° Impulse noise (produced in a subscriber loop plant) Narrowband interference (picked up from a nearby AM radio station) Discrete multitone (DMT) system The DMT receiver spreads the energy of an impulse over many subchannels, thereby reducing its degrading effect Due to the sinc(x) spectral characteristic of each subchannel, the DMT receiver tends to pass the interference into. many subchannels. Since the signal power in each subchannel is only 1/N of the total signal power, susceptibility of the receiver to narrowband interference can be acute Carrierless amplitude/phase (CAP) modulation system ‘The CAP system relies on the use of a single carrier, It is therefore susceptible to impulse noise. (This problem may be mitigated by the use of a powerful code such as the Reed-Solomon code.) The use of an adaptive equalizer is based on the mean-square error criterion. The effect of narrowband interference can therefore be reduced by creating a noteh in the receiver performance at the frequency of the interferer. The comparative p bibliography Problem 6.46 InMx -y FSK, the transmitted signal is defined by 369 ts made in the table presented herein are based on Saltzberg (1998); see the300 = [Ecos Bin, +1), O>I, the second term on the right-hand side of Eq. (3) has a negligible effect; hence, this modulation scheme has an average probability of symbol error practically the same as that for coherent QPSK or MSK. For the coherent detection of differentially encoded QPSK, the average probability of symbol error is given by E, o [E 1. af [E,) 2erfe) || —2erfe"| |) + erte = gerfe!) | ( i] (fe (Wie) 47° Ly 371For large £)/No, this average probability of symbol error is approximately twice that of coherent QPSK. Problem 6.48 (a) Assuming that the input data sequence x(n] is measured in volts, and recognizing that the symbol 4, is dimensionless, then from Eq. (6.271) we find that the error signal e[n] is also measured in volts. Then, with the phase estimate 6[n] measured in radians, it follows that the step-size parameter for carrier recovery in Eq. (6.272) is measured in radians/volts. (b) From Eq. (6.282) defining the error signal in terms of the input data sequence, we see that the error signal is measured in volts squared. Hence, with c[n} responsible for timing recovery, measured in seconds, it follows that the step-size parameter y for timing recovery in Eq. (6.286) is measured in seconds/volts”. Problem 6.49 (a) The complex envelope of the received waveform is given by F(t) = 3(t) + W(2) dd Il where 5(1) = ?™* Sy ayo(r—KT 1) i and (i) is the channel noise. The parameter v represents the frequency offset, @ is the carrier phase we want to estimate, t is the timing error, {a,} is the sequence of information symbols, Tis the symbol period, and g(t) is the signaling pulse shape. ‘The likelihood function L(r|8) is given by T) To ! [Retrosena = ay, J Beara @ 0 Since |3(f)] is independent of the carrier phase @ , the log-likelihood function of @ is given by 372T,) 1(8)=log(L(r|8)) = Re: frox'wal oD) ° j 1%, fol where J 7038" (Ndt = eS ajx(k) ° i where x(k) represents the sample taken at time f= AT +t in the formula for convolution: = pave xM=ir(e 1 *a(-1) Therefore, ll fer > «isco | k=0 (8) = ‘The maximum of /(0), i.., maximum likelihood estimation of @, is achieved for to" 6 = arg) Lain} (4) (ico (b) Hence, from Eq. (4) we deduce the following system for estimating the phase 0 (k) A +{_) my = ® + 2 of arg} }» 6 tktee 373Problem 6.51 Matlab codes u Problem 6.61 a. CS: Haykin Yettect of a dispersive channel on BPSK signals %M. Sellathurai 4% number of bits and number of samples per bit no_of_syms no_of bit: % generating bits Bits=[1 10110400417; 4% generating QPSK signals [oyms]=BPSK_mod(no_of_bite, Bits); tsH1e-3/16; A=Length (eym 4% baseband signal s=zeros(sanples_per_bit*(1) ,1); for ket:1-1 for kk=0: (samples_per_bit-1) ((k-4)*samples_per_bittkk+t,1)=syns (1k); end end ts: (Length(s)-1)#ta; % channel bandwidths 28-12, filter order 2N=10 N=5; H12-butter_channel (2+B,N); TT12=conv(Hi2, »: % channel bandwidths 21 6 filter order 2N=10 Hi16=butter_channel (2+B,N); ‘TT16=conv(H16, 8); 374% channel bandwidth 242-20 filter order 2 H20=butter_channel (2+8,N); TT20=conv(H20, s); Yehannel bandwidth 248=24 filter order 2 25 Ho4=butter_channel(2+B,); TT24=conv(H24, 8); % channel bandwidth 2 85 430=butter_channel(2*B,N); ‘TT30=conv(H30, 8); 10 filter order 2N=10 % prints subplot (2,3,1) hold on for k=1:10 plot (ke, Leyms(ie)]’,%0”) Line((k, 1, [0 syms(k)) end xlabel(’Bit period’); title( ‘Transmitted bits’); hold off subplot (2,3,2) Cm start}=max(real(12)); hold on plot(t,s,’--")3 plot(t,real (TT12(start: 160+start-1))); xlabel (time (s)' title( Baseband BPSK, BY=12kHz"); hold off subplot (2,3,3) Un stert]-max(real (Hi6)); hold on plot(t,s,’——'); plot(t,real (IT16(start: 160+start-1))); xlabel(’tie (s)'); title(’ Baseband BPSK, BV=i6kHiz’); hold off 375subpiot(2,3,4) Um start] =max(real(H20)); hold on plot(t,2,?——); plot(t,real(TT20(start :160+start-1))); xlabel(’time (s)'); title(’Baseband BPSK, BW=20kHz'); hold off subplot (2,3,) Gn start]=max(real (H24)); hold on plot(t,s,'--"); plot(t,real (TT24(start:160+start-1))); xlabel('tine (5)'); title(Baseband BPSK, BY=24¢Ki"); hold off subplot (2,3,6) Gm start]-max(real(H30)); hold on plot(t,s,?=-"); plot (t,real (7730(start: 160+start-1))); xlabel (time (s)’); title( "Baseband BPSK, BY=30KH2"); hold off 376% Problem 6.51 b. CS: Haykin Yettect of a dispersive channel on QPSK signals %M. Sellathurai % number of bits and number of samples per bit no_of_syne no_of_bits=no_of_syms*2; samples_per_bit=i6; generating bits ‘ound (rand(no_of_bits, 1)); 4% generating QPSK signals [syns] =9PSK_mod(no_of bits, Bits); ength(syms) ; % baseband signal eros (samples_per_bit*(1-1),1); for k=1:1-1 for kk=0: (samples_per_bit-1) 8((k-1)#samples_por_bit+kkt1, 1)=syns(1,k); end end t=0:te: (Length(s)-1)4t8; 2, filter order 21 channel (248) ; TT12-conv(H12, #); % channel bandwidths 28-16 filter order BB; H16-butter_channel(2+B,N); ‘TT16=conv(Hi6, 8); 4% channel bandwidth 2*B-20 filter order 24-10 itter_channel (24B,N); onv(H20, 5); 377Wehannel bandwidth 2#B=24 tilter order 2N=10 H24=butter_channel (2+B,4); TT24=conv(Hi24, 8); % channel bandwidth 248-30 filter order 2N=10 B15; H30=butter_channel(2+B,N); TT30=conv(H30, 8); % prints subplot(2,3,1) hold on for k=1:10 plot(k, [2#Bits(k)-1]’, 0") Lime(Dk, kK), [0 (2¥Bits(k)-1)1) end xlabel(’Bit period’); title( Transmitted bits’); hold off subplot(2,3,2) Cm start}=nax(real(Hi2)); hold on plot(t,s,’--'); plot (t,real (TT12(start:64+start-1))); xlabel('time (#)’); vitle(’Baseband QPSK, BY=i2kiiz’); hold of¢ subplot(2,3,3) [m start]-max(real (Hi6) ); hold on plot(t,s,?=-"); plot (t,real (1T16(start:64+start~1))); xlabel ("time (s)’); title(’ Baseband QPSK, BW=i6kliz’); hold off subplot(2,3,4) Gm start]=max(real (H20)); hold on plot(t,s,!=-"); plot(t,real(TT20(start:64tstart-1))); 378xlabel (time (s)’); title(’Baseband QPSK, BY=20kH2’); hold off subplot(2,3,5) [m start]=max(real(H24)); hold on plot(t,s,’=-"); plot(t,real (TT24(start:64+start-1))); xlabel(’time (8)’); bitie(’Baseband QPSK, BW=24KH2’); hold oft subplot(2,3,6) Um start}=max(real(H30)); hold on plot(t,s,’--'); plot(t,real (TT30(start:64+start-1))); xlabel (time (s)’); title( Baseband QPSK, BY=30Kiiz"); hold off 3794 Problem 6.51 c. CS: Haykin Yettect of a dispersive channel on ¥SK signals %M. Sellathurad % number of bits and number of samples per bit no_of_syme no_of _bits=no_of_syms*2; sanples_per_bit=i6; 4% generating bits $101101000)%; 4% generating QPSK signals (s,phase]=NSK_nod(no_of_bits,samples_per_bit,Bits); % channel bandwidths 2: 2, filter order 2N=10 Hi2-butter_channel (2¥B,N); TTi2=conv(H12, 8); % channel bandwidths 28-16 filter order 2 jut ter_channel (2#8,N); TT16=conv(H16, 8); % channel bandwidth 24 B=10, H20=butter_channel (2#3,M); ‘TT20=conv(H20, 8); 0 filter order 2N=10 Ychannel bandwidth 2¢i B=12; H24=butter_channel(2+B,); ‘TT24=conv(H24, 8); 4 filter order 2N=10 % channel bandwidth 268-30 filter order 2N=10 Be15; aN); TT30=conv(H30, s); 1e-3/16; ts: (Length(s)-t)+ts 380subplot(2,3,1) hold on for k=1:10 plot(k, [2*Bits(k)-1]',’0%) Line(Ce, KJ, [0 (2¢Bits(k)-1)]) end xlabel(’Bit period’); title( Transmitted bits’); hold off subplot (2,3,2) Cm start]=nax(real(#i2)); hold on plot(t,abs(s),!--"); Plot (t, abs(TT12(start+s: 165¢start-1))); xlabel(*time (8)’); title(MSK (envelope), BW=12kiz"); hold off axia((0, 0.01,0.9,4.1]) subplot (2,3,3) (m start}-nax(real (iii6)); hold on plot(t abs(s),"-—"); plot(t,abs(1716(start+S:165+start-1))); Mlabel( tine (s)’); title? HSK (envelope), BY=LeKH2" hold off axis({0, 0.01,0.9,1.1 1) subplot (2,3,4) (m start}=max(real(H20)); hold on plot(t,abs(s),?—-"); Plot(t,abs(TT20(start+6:165+start-1))); xlabel(’tine (s)'); title(MSK (envelope), BW=20kH2"); hold off axis((0, 0.01,0.9,1.1 3) subplot(2,3,5) (m start]=max(real (24); hold on PLot(t,abs(s),'--); plot(t,abs(TT24(start+s:165+etart-1))); 381xlabel(’time (s)'); title(’MSK (envelope), BW=24kHz"); hold off axis([0, 0.01,0.9,1.1 J) subplot (2,3,6) (m start]=nax(real(H20)); hold on Plot(t,abs(s),’--"); plot(t(1:185) ,abs(TT30(start+s: 160+start-1))); xlabel(’time (s)’); title(MSK (envelope), BY=30KH2"); hold off axis([0, 0.01,0.9,1.1]) 382% Problem 6.51 d. CS: Haykin Wettect of a dispersive channel on GMSK signals 4M. Sellathurai % number of bits and number of samples per bit no_of_syne no_of bits no_ot_syms#2; sanples_per_bit=16; % generating bits Bits-[1104101000]%; 4% generating GNSK signals, WTb=0.3 [s, phase]=GNSK_nod(no_of_bits,samples_per_bit,Bits); 4% channel bandwidths 28-12, filter order 2Ni=10 ut ter_channel(2+B,W); nv (HA2, 5); % channel bandwidths 21 6 filter order 2N=10 H16-butter_channel (2+B,¥); TT16=conv(Hi6, 5); 4% channel bandwidth 2+ 0; H20=butter_channel(2+B,N); ‘T120=conv(HH20, 8); 90 filter order 2=10 Yehannel bandwidth 2¢i B12; H24=butter_channel(2¥B,4); ‘TT24=conv(H24, 8); 4 filter order 2N=10 channel bandwidth 2*B=30 ¢ilter order ‘t=0:t8: (Length(s)-1)+ts 383% prints subplot(2,3,1) hold on for k=1:10 plot(k, [2*Bits(k)-1]’,'0’) Line(Ek, kK), [0 (2*Bits(k)=1)]) end xlabel (Bit period’); title( Transmitted bits’); hold off subplot(2,3,2) Cm start]=max(real(Hi2)); hold on plot(t,abs(s),’--"); plot(t,abs(TT12(start: 160+start-1))); xlabel ("time (s)’); title(?GHSK (envelope), BY=12kii2"); hold off axis([0, 0.01,0.9,1.1) subplot (2,3,3) Ue start]=nax(real(H16)); hold on plot(t,abs(s),'=-"); plot (t,abs(TT16(start: 160+start-1))); xlabel (time (s)’); title(? GHSK (envelope), BYS16kH2"); hold off axis((0, 0.01,0.9,1.1 1) subplot (2,3,4) [m start]=max(real (H20)); hold on plot(t,abs(s), =~"); plot(t,abs(TT20(start: 160+start-1))); xlabel(’time (s)’); title(’GHSK (envelope), B¥=20ki2’); hold off axis([0, 0.01,0.9,1.1 ]) subplot (2,3,5) {Un stare}max(reai(i24)); hold on plot(t,abs(s), 7); plot(t,abs(1T24(start:160+start-1))); 384xlabel( ‘time (s)’); title(’'GMSK (envelope), BY=24kHiz’) ; hold off axis((0, 0.01,0.9,1.1]) subplot(2,3,6) [n start]=max(real (H30)); hold on plot(t,abs(s),?=--"); plot (t,abs(TT30(start : 160+start-1))); xlabel (time (s)’); title(’GMSK (envelope), BY=30kH2”); hold off axis([0, 0.01,0.9,1.1 1) 385function [amp]=BPSK_nod(no_of_bits, b) % used in problem 6.51(a), CS: Haykin % BPSK modulation % Mathini Sellathurai elseif (b(k amp(1)= 1; ond Ltt end 386function [amp]=QPSK_nod(no_of_bite, b) 4% used in problen 6.51(b), CS: Haykin 4% QPSK modulation 4% Mathini Sellathurai amp=01; if (b(K)==0 & bleet) == 0) anp(2)= (-1444-1)/sqrt(2); elseif (b(K)==1 & b(kH) amp(1)= (1-i#1)/sqre (2); elseif (b(k)==1 & b(k+1) amp(1)= (1#ie1)/sqrt(2) else (b(e)==08 ber) amp(a)= (-1+i#1)/sqrt(2 ) end delet endfunction [amp,phase]=4SK_mod(no_of_bits, samples_per_bit, b) % used in problem 6.51(c), CS: Haykin % MSK signal generator % Mathini Sellathurai anp=(; 4ni_phase: for ket:no_of_bits e0rb(xe); for kk=0:samples_per_bit-1 ¥%NRZ signal generator i_phase 4e0*(pi/(2+samples_per_bit)); ini _phase=phase( (x-1)+samples_per_bittkk+1); amprintiequad; 388function [anp, phasei]=GUSK_mod(no_of_bits, samples_per_bit, b) 4% used in problem 6.51(4), CS: Haykin % GUSK signal generator o Mathini Sellathurai amp= Oj tor k :no_of bits ‘Usenerating RZ sequence it b(k,1): in_bite(k, 1): else im_bits(k,1) end end impulse_bits=im_bits; Bits_to_transmit=nax(size(impulse_bits)); BT=0.6; inphase=0; @ata(1,4)=0; 20; tor :3 for (samples_per_bit -1) co =GNSK_co(i-2,k+8, samples_per_bit BT); uskcoef (1, i+samples_per_bit+k+i)=co; end end for bitcount=1:Bits_to_transmit ini_phase=inphase; ini_phasesren(ini_phasetdata(1,4)*pi/2,2¢pi); data(t,1)=inpul: for i =4:n4:2 data(t,i)=data(1,i-1); end sits (bitcount, 1); inphas ini_phase; for pha_loop=1: samples_per_bit. phase=inphase; for i=0:3 phase=phase+pi/2+data(1, i+1)+qmskcoef (1,samples_per_bitwitpha_loop) ; 389)store(1,t+1): phase=rem(phase,2#pi); phasei(1, pha looprsamples_per_bite(bitcount-1))=phase; Tephase(1.pha loop+samples_per_bit+(bitcount-1))=cos (phase) quphase(1,pha_loop+samples_per_bit+(bitcount~1))=sin(phase) ; ond end amp=rephaset}+quphase; 390function [co] = GHSK_co(a, b, samples_per_bit,bt) % used in GUSK signal generation Problem 651d, CS: Haykin % Mathini Sellathurai alpha=bt+5. 336446225; ‘Tea+b/samples_per_bit; co=Teors (T+alpha)+exp(-alphatalphaeTeT) /(alpha #1.772453655) ; co=con(T-1) sext(((T-1) ¢alpha)~exp(-alphatalphae (T-1)#(T=1))/(alphassqrt(pi)); 5+0..58co; 391function hb=butter_channel(f,N) % Used in Problen 6.51 % Butterworth filter of order 2N=10; %M, Sellathurai (B, Al-buttor(, £/64); [Hw] =freqz(B,A, 128, whole’); mbaizee (3); 392Answer to Problem 6.51 ne (2) x0" Figure |: Butterworth baseband filter of order N-5 393roe Transmitted bits Baseband BPSK, BW=12kHz Baseband BPSK, BW=16kHz 1 15) 15) 1 1 : y {Y 05: | fee 08 0s 0 0) 0 0.5 0.5 0.5 t I “1 a h i ~ 15 15 oO 5 10 0 0.005 0.01 0 0.005 0.01 Bit period time (s) time (s) Baseband BPSK, BW=20kHz Baseband BPSK, BW=24kHz Baseband BPSK, BW= 15, 15) 1 1 08: 0s 0 0 -05 08 a 15 15 0 0.005 0.01 0 0.005 0.01 oO 0.005 0.01 time (s) time (8) time (s)Transmitted bits Baseband QPSK, BW=12kHz Baseband QPSK, BNV=16kKHz 1 1 1 \! j \ 1 \ 05 05 q i 05 q ! " q 0 ol 0 { ' q hook i -05 -05 \ ft 0.5 ' ! 1 0 8 10 2 4 2 4 Bit period time(s) 49°? time(s) x 407° Baseband QPSK, BW=20kHz Baseband QPSK, BW=24kHz Baseband QPSK, BW=30kH2 1 1 1 1 \ 1 os} |i ost | o.5| q I ° ° ° I h h 0.5) i 05 I" 0.5 \ f 2 4 2 4 ~o 2 4 time(s) 49? time (8) 49°? time(s) 40°Transmitted bits MSK (envelope), BW=12kHz MSK (envelope), BW=16kHz 1 14 14 05 1.05 4.05 o 1 1 -05 | 0.95 0.95 os) 09 e 5 10 0 0.005 oor "0 0.005, 0.01 5 Bit period time (s) time (8) MSK (envelope), BW=20kHz MSK (envelope), BW=24kHz MSK (envelope), BW/=30kHz 14 44 14 1.05 1.05, 1.05 1 1 1 0.95 0.95 0.95 os! 09 09 0 0.005 001 "0 0.005) 001 "0 0.005 0.01 time (6) time (s) time (s)Transmitted bits GMSK (envelope), BW=12kHz GMSK (envelope). BW=16kHz 1 14 1 os 1.05| 1.08 0 1 1 8 “o 5 70 0 0.005 oor °° 0.005 0.01 S Bit period time (s) time (s) | OMSK emap) Bwiezo GSK (ene), Be2He MEK (enetore BME: 4.05 1.08 1.05 1 1 1 0.95 o.9s| 0.95 °. %0 0.005 0.01 t: % 0.005 0.01 08 0.005 0.01 time (s) time (s) time (:)‘et penoa! Figure 2 ‘Transmitted bits sk peso ve) Figure 3 Phase of baseband MSK signal 398me) Figure 4 T and Q components of baseband MSK signal Figure § Phase of bascband GMSK signal 399Figure 6 I and Q components of baseband GMSK signal 400CHAPTER 7 |-spectrum, Problem 7.1 (a) The PN sequence length is (b) The chip duration is tT, = 1s -01ns 107 (c) The period of the PN sequence is T=NT, 15x 0.1 = 15 ns Problem 7.2 Shift Shift-register Modulo-2 Shift-register number contents adder output output 0 1000 1 0100 0+0=0 0 2 0010 0+0=0 0 3 1001 0 4 1100 O+l=1 1 5 0110 0+0=0 0 6 1011 1+0 0 7 0101 1+1=0 1 8 1010 O+1=1 1 4019 1101 14021 0 10 1110 O+1=1 1 ll 1111 1+0=1 0 12 0111 14+1=0 1 13 0011 1+1=0 1 14 0001 14+1=0 1 15 1000 O+1=1 a The output sequence is therefore 11 000100110101111 0001 one period Problem 7.3 (a) From both Table 7.2a and Table 7.2b we note the following: Balance property: Number of 1s in one period = 16 Number of 0s in one period = 15 Hence, the number of 1s exceeds the number of 0s by one. (b) Run property: In both Tables 7.2a and 7 2b, we count a total of 8 runs of 1s and a total of 8 runs of 0s, Moreover, we note the following: Runs of length 1: 4 Runs of length 2: 2 Runs of length 3: 1 (©) Autocorrelation function: olHence, we have (not to seale) 6; 403Problem 7.4 Se L—{_s |+4 Ls } +4 Table 1 Feedback State of Feedback- Output symbol Bor eEHoOnHooHnHeOHOoOHOHH HHH shift register 100000 110000 111000 111101 ee ee 111111 011111 101111 010111 101011 010101 101010 110101 011010 001101 100110 110011 011001 101100 110110 111011 011101 101110 symbol BE OoOH HOH OHOHH HHH HOO OOO 404Table 1 continued Feedback symbol . COP OH C OPH HH OH CCOHHH OOH OOH ON State of feedback- shift register 110111 011011 101101 010110 001011 100101 010010 001001 100100 110010 111001 011100 001110 000111 100011 010001 101000 110100 111010 111101 011110 001111 100111 010011 101001 010100 001010 Output symbol Bee enon CoCOH HH OOH OC OH OHH OHHH O 405Table 1 continued Feedback State of feedback- Output symbol _shift register symbol 000101 100010 110001 011000 001100 000110 000011 100001 010000 001000 000100 000010 000001 100000 HeccooHocooHHS HeoocoHooecoHoHeE 406Table 2 6 [—¥ aR Feedback State of feedback- Output symbol CH HH HHH OH OHHH OCCOHRHOCOHHH ON shift register 100000 110000 011000 101100 110110 111011 011101 001110 100111 110011 011001 001100 000110 100011 110001 111000 011100 101110 010111 101011 110101 111010 111101 111110 111111 [Ue Ue Ge ia es symbol HOH OHHH OSC OHH OCH HH OHH COO O 407Table 2 continued Feedback State of feedback- Output symbol __ shift register symbol 101111 110111 011011 101101 010110 001011 000101 100010 010001 001000 000100 000010 100001 010000 101000 110100 011010 001101 100110 010011 101001 010100 101010 010101 001010 COP SCH OH CORN SH CC OCCH OCC OH OHH HROoOHH OH OSCO OH SCC OH OHH OHH HETable 3 Feedback symbol COoOPHoCeC OH OH OH OOH H HH HHO ¥ State of feed-back shift register 100000 010000 101000 110100 111010 111101 111110 111111 011111 001111 100111 010011 101001 010100 101010 010101 001010 000101 100010 110001 011000 001100 aa Output symbol CHOH CHOC OHHH HHH OH OSC OOO 409 {os oHTable 3 continued Feedback State of feedback- Output symbol __ shift register symbol 100110 110011 111001 111100 011110 101111 110111 111011 011101 101110 010111 101011 110101 011010 101101 010110 001011 100101 110010 011001 101100 110110 011011 001101 000110 SSCHH OHH OOH OHH OH OHHH OHM HE BPR OCH OMH OH OHHH OM M HHO OHHOSTable 3 continued Feedback State of feedback- Output symbol _ shift register symbol 1 100011 0 010001 0 001000 1 100100 0 010010 0 001001 0 000100 0 000010 1 100001 1 110000 1 111000 0 011100 0 001110 0 000111 0 000011 0 000001 a 100000 Hen ScocoecHoOOoHOoOoHHOTable 3 continued Feedback State of feedback Output symbol _shift register symbol 100101 010010 001001 100100 110010 111001 111100 011110 oo1111 000111 000011 000001 100000 HecoeooHH HH OOH HH HH oOoOH OC OHOHO 412State of feedback- Output shift register symbol Initial stat 10000 01000 00100 00010 00001 11101 10011 10100 01010 00101 11111 10010 01001 11001 10001 10101 10111 10110 01011 11000 01100 00110 00011 11100 01110 00111 11110 01111 11010 01101 11011 10000 BH oOHOHCOH OCC OH OHH HH HOH HOC OH HH OCC O 413The 31-element code generated by the scheme shown in Fig. P9.2 is exactly the same as that described in Table 9.2b. Note, however, the code described in Tab.e 9.2b appears in reversed order to that described in the above table; this reversal is clearly of a trivial nature. 414Problem 7.6 (a) The modulo-2 sum of b(t) and c(t), on a pulse-by-pulse basis, is as follows bw 1 e(t) 0 0 1 1f1 0 (b) If symbol 0 is represented by a sinusoid of zero phase shift, and symbol 1 is represented by a sinusoid of 180° phase shift, the output of the modulo-2 adder takes on the same form as that described in Table 7.3 of the text. Problem 7.7 itt) = V2T cos(2nfyt + 8) The basis functions are ato = cos(2nf,t), kT, St < (k+DT, c 0, otherwise Ac = sin(2nfct), kT, oO Ld sw (Ginary dats) Baseband ‘ler co) ‘The DS/QPSK modulated signal is |E |E x(t) (1)cos(2n f.1) + Feots)sin(2nf.t) where eft) = {¢9, (0, €4, (ss Cyn, CE} and Colt) = {ep, (4.61, ols ey-4, of} denote the spreading sequences for 0 <1 < T,, which are applied to the in-phase and quadrature channels of the modulator. Consider the following set of orthonormal basis functions: os(2nft), kT S1S(k+1)T, otherwise 418i [Fsmcns.n, ) AT ,StS (k+1)T, 0, otherwise Go, = where T, is the chip duration; & = 0, 1, 2, «++, N-1,and N= TIT,, that is, N is the number of chips per bit ‘The DS/QPSK modulated signal can be written as follows (using the set of basis functions): T 2 TE [2 a [Pooenranetns oF Fonens.ncac [25 ze. 190, 4, {2 Xo, 1% eg ‘The channel output at the receiving end of the system has the following form A(t) = s(t) + f(D) where j(t) denotes the interference signal. We may express the interference signal using the 2N-dimensional basis functions as follows: wa xa FO = Yer ie, + Dieg. (eg (0 k= k=O " where (EN, Jin, "HO%, (D4 Jeng (kstyr jean = Sur, HOMeg Oat k= OLN The average power of the interferer is given by 419Assuming that the power is equally distributed between the in-phase and quadrature components: N-1) Sees ee L Matis sie ef) There are two stages of demodulation. First, the received signal x(7) is multiplied by the despreading sequences et) and co(0), yielding u(t) uo(t) Feos(2m ft) £ exneo(Esint2n7.0 +e(O() Esnensn t+ eqhne so) Boosters. + eo(t)i(t) 420The second terms in the right-hand side of uj(t) and ug(t) are filtered by the bandpass filters, and the BPSK demodulators recover estimates of their respective binary sequences. Finally, the multiplexer reconstructs the original binary data stream, Proc: it The signal-to-noise ratio at the output of the receiver is (SNR), = Mstantaneous peak signal _power ‘The signal-to-noise ratio at the input of the coherent receiver is average input-signal power (SNR), average interferer power We may therefore write (SNR) oi 2: (r oe, GAR, = totes, (7) = 3+ 10log, | 7) The QPSK processing gain= T/T, That is, PGaesx = 21PGypsx] Solving for the antenna aperture: ziProblem 7.12 ‘The processing gain (PG) is PG Hence, expressed in decibels, PG = 101og1920 = 26 db Problem 7.13 The processing gain is PG =4x4 = 16 Hence, in decibels, PG = 101ogi916 = 124B 422Problem 7.13 Matlab codes % Problem 7.13(a), CS: Haykin 4% Generating 63-chip PN sequences % polynomial (x) = x°6+x+1 4% polynomial2(x) = x°6 + x°6 + x24 x44 4% Mathini Sellathurai, 10.08.1999 % polynomials polt={t 0000143; polz[1 100414]; % chip size 35 % generating the PN sequence pnseqi = PNseq(pol1); pnseq? = Pliseq(pol2); % mapping antipodal signals (0-->-1, 1-->4) u=2epnseqi~1; spnseq?-14% autocorrelation of pnseqi [eorrf]=pn_corr(u, u, M) % prints plot(~61:62,corrt(2:126)); axis([-62, 62,-10, 80]) xlabel(’ Delay \tau’) ylabel(’ Autocorrelation function R_{c}(\tau)’) pause Yautocorrelation of pnseq? [eorr#]=pn_eorr(v, v, N) % prints plot (~61:62, corrf(2:125)); axis([-62, 62,-10, 80]) xlabel(? Delay \tau’) ylabel(? Autocorrelation function R_{ch(\tau)’) pause 4% cross correlation of pnseqi, pnseq? [eucorr]=pn_corr(u, v, W) % prints plot (-61:62,c_corr(2:426)); axis([-62, 62,~20, 20]) xlabel(’ Delay \tan’) ylabel(? Cross-correlation function R{Ji}(\tau)’) 4244% Problem 7.13 (b), CS: Haykin 4% Generating 63-chip PN sequences % polynomiali(x) =x6+x+4 % polynomial2(x) = x6 + x6 + x24 x41 % Mathini Sellathurai, 10.05.1999 % polynomials polis{i 1100141; pola[t 1001147; % chip size N63; % generating the PN sequence pnseqi = PNseq(pol1); Pnseq? = PNseq(pol2); % mapping antipodal signals (0-->-1, u=2+pnseqi-1; v=2epnseq2-1} 4% autocorrelation of pnseqi (corr#]=pn_corr(u, u, MN) % prints plot(-61:62,corrt(2:126)); axis([-62, 62,-10, 80]) xlabel(’ Delay \tau’) ylabel(’ Autocorrelation function R{c}(\tau)’) pause autocorrelation of pnseq2 Ceorrf}=pncorr(v, v, N) 4 prints plot(-61:62,corré(2:125)); axis([-62, 62,-10, 80]) xlabel(? Delay \tau’) ylabel(? Autocorrelation function R_{c}(\tau)") pause % cross correlation of pneegi, pnseq? (e_corr]=pn_corr(u, v, ND % prints 425plot(~61:62,c_corr(2:426)); axis([-62, 62,-20, 20)) Habel(? Delay \tau’) ylabel(’ Cross-correlation function R{ji}(\tau)’) 426function x = Plseq(p) 4% Linear shift register for generating PN sequence of polynomial p % used for problens 7.13, 7.14 of CS: Haykin % Mathini Sellathurai, 10.08.1999 Length(p) - 1; % order of the polynomial fLipir(p); (1 zeros(1, N-1)]; 45 " P x for i= 1: ne(2-N- 1) x(4) = XC); X= [X(2:m) p(t) * rem(sum(p(1:W) .* x(4:N)), 20]; ond 427o function [corrf]=pn_corr(u, v, N) % funtion to compute the autocorreation/ cross-correlation % function of two PN sequences % used in problem 7.13, 7.14, CS: Haykin 4 Mathini Sellathurai, 10 june 1999. for m=0:11 shitted_u=[u(m+1:N) u(1:m)]; corr(m+1)=(sum(v.#shifted_u)); if (abs(corr)>max_cross_corr) max_cross_corr=abs(corr) end end 428Answer to Problem 7.13 180 pe Figure: Autocorrelation function of [6,5,2,1],6,1) 429eo = = obs 3 we 70 Figure 2: Cross-correlation function of [6,5,2,1] 6,1] beiays ire J: Cross-correlation function of (6,5,2,1]{8,5,4,1] 430Problem 7.14 Matlab codes 4% Problem 7.14 (a), CS: Haykin 1 Generating 31-chip PN sequences % polynomiali(z) = x76 + x°2 +1 % polynomial2(x) = x75 + x73 +4 % Mathini Sellathurai, 10.05.1999 % polynomials poli=[10010 17; 101004); % chip size N31; 4% generating the PN sequence pnseqi = PNseq(polt); pnseq? = PNseq(pol2): 4% mapping antipodal signals (0-->-1, ue2epnsegi-1 v-2epnseq2-1 % cross correlation of pnseqi, pnseq2 [e_corr]=pn_corr(u, v, N) % prints plot(-30:31,c_corr); axis([-30, 31-18, 18]) xlabel(? Delay \tau’) ylabel(? Cross~correlation function R_{ji}(\tau)*) 4314% Problem 7.14 (b), CS: Haykin 4% Generating 63-chip PN sequences 4% polynomiali(x) = x75 + x73 +4 4% polynomial2(x) = x°6 + x74 + De KHL 4% Mathini Sellathurai, 10.05.1999 4% polynonials % chip size N31; 4 generating the PN sequence pnseqi = Piseq(polt. phseq? = Pliseq(pol2); % mapping antipodal signals (0-->-1, epnseqi-1; epneeq2-1; 4% cross correlation of pnseqi, pneeq? (c_corrl=pn_corr(u, v, ¥) % prints plot(-30:31,c_corr); axis([-30, 31,-10, 10]) xlabel(? Delay \tau’) ylabel(? Cross-correlation function R{ji}(\tau)’)% Problem 7.14 (c), CS: Haykin % Generating 63-chip PN sequences % polynomiali(x) = x°6 + x-4 + x-3tt % polynomial2(x) = x6 tx4+x2+x+1 % Mathini Sellathurai, 10.05.1999 % polynomials 1111042; 104110; % chip size N34; 4% generating the PN sequence pnseqi = PNseq(polt); phseq? = PNseq(pol2); % mapping antipodal signals (0- ue2¢pnseqi-1; v=24pnseq?-1; % cross correlation of pnseqi, pnseq2 {e_corr]=pn_corr(u, v, N) % prints plot (-30:31,c_corr); axis([-30, 30,-10, 10]) xlabel(’ Delay \tau’) ylabel(? Cross-correlation function R_{ji}(\tau)’) 433Answer to Problem 7.14 Casco Figure {: Cross-correlation function of [5,3}[5.2] 434lint . Delay Figure 3: Cross-correlation function of {6,5.2,1],[6,5,4.1] 435Problem 8.1 2 (a) Free space loss = 10log (42 wh A 4xnx 150 = 20log [—~7=—= | 103 x 10°74 x 10” JaB = 88 dB (b) The power gain of each antenna is LOlog gG, = 10log yG, = 10log (* 10 = 1log (tnxn<06) ws (3/40) = 36.24 dB (©) Received Power= Transmitted power +G, - Free space loss = 1+36.24-88 =-50.76 dBW Problem 8.2 The antenna gain and free-space loss at 12 GHz can be calculated by simply adding 20log,9(12/4) for the values calculated in Problem 8.1 for downlink frequency 4 GHz. Specifically, we have: (a) Free-space loss= 88 + 2010g30(3) = 97.54 dB (b) Power gain of each antenna 36.24 + 2010g19(3) = 45.78dB (c) Received power = -50.76 dBW ‘The important points to note from the solutions to Problems 8.1 and 8.3 are: 1. Increasing the operating frequency produces a corresponding increase in free-space loss, and an equal increase in the power gain of each antenna. 2. The net result is that, in theory, the received power remains unchanged. 436Problem The Friis free-space equation is given by (ey P= POG (a) Using the relationship a) Q) In both Eqs. (1) and (2) the dependent variable is the received signal power, but the independent variables are different. (©) Equation (1) is the appropriate choice for calculating P, performance when the dimensions of both the transmitting and receiving antennas are already fixed. Equation (1) states that for fixed size antennas, the received power increases as the wavelength is decreased. Equation (2) is the appropriate choice when both A, and G, are fixed and the requirement is to determine the required value of the average transmitted power P, in order to realize a specified P, Problem 8. The free space loss is given by 4374nd)? L = (Se sonee=(‘84) According to the above formulation for free space loss, free space loss is frequency dependent. Path loss, as characterized in this formulation, is a definition based on the use of an isotropic receiving antenna (G, = 1). ‘The power density, p(d), is a function of distance and is equal to EIRP (d) 3 4nd’ The received power of an isotropic antenna is equal to IRP/ free-space w Equation (1) states the power received by an isotropic antenna is equal to the effective transmitted power EIRP, reduced only by the path loss, However, when the receiving antenna is not isotropic, the received power is modified by the receiving antenna gain G,, that is, Eq. (1) is multiplied by G, Problem 8.5 In a satellite communication system, satellite power is limited by the permissible antenna size. Accordingly, a sensible design strategy is to have the path loss on the downlink smaller than the pass loss on the uplink. Recognizing the inverse dependence of path loss on the wavelength A, it follows that we should have 2upkink < Adowatink or, equivalently, Foplink > Faowntink 438Problem 8.6 Received power in dBW is defined by Pr=EIRP + G,-Free-space loss a) For these three components, we have (1) EIRP " 10log ,(P,G,) 10log ipP, + 10log ,4(G,) 10log jo(0.1) + 10log ,(G,) Q) Transmit antenna gain (in dB): 4x2x0.7x 2/4) 10log yG, = 10log " e/a 7 = 30.89 dB 8) (2) Receive antenna gain: ( lOlog pG, = 1log | 4XEXOSSx ax 5” 10 (3/40) = 49.84 dB 4) (3) Free-space loss: , (AXRX 4) oat aes 4x 1x 4x10 = 2 4xmx 4x10) M08 3740 = 196.25 dB 6) Hence, using Eqs. (1) to (5), we find that 439P,, = 10logio(0.1) + 30.89+49.84 ~ 196.52 0 ~ 206.52 + 8.073 -125 dBW Problem 8.7 (a) RMS value of thermal noise = JAKTRAF volts, where k is Boltzmann's constant equal to 1.38 x 107, Tis the absolute temperature in degrees Kelvin, and R is the resistance in ohmns. Hence, RMS value= 4 x 1.38 10° x 290 x75x1x 10° = lax 1.38 x 29075 x 10-7 = 1.096 x 10° volts (b) The maximum available noise power delivered to a matched load is KTAf = 1.38% 10 x 290x 10° = 4.0x 105 watts Problem 8.8 The wavegude loss is 1 dB; that is, Gwaveguide = 0-78 The noise temperature at the input to the LNA due to the combined presence of antenna and waveguide is Te = Gvaveguide * Tantenna * (1 ~ Gyavepuide)T waveguide = 0.78 x 50 + 290(1 - 0.78) = 102.8K The overall noise temperature of the system is 440500 , 1000 Teysiem = Te + 50+ 505 + 9G" = 160.3K ‘The system noise temperature referred to the antenna terminal is 160.3/0.78 = 205.5K Problem 8,9 In this problem, we are given the noise figures (F) and the available power gains (G) of the devices. By using the following relationship, we can estimate the equivalent noise temperature of each device: poltte 7 T, = T(F-1) where T is room temperature (290K) and , is the equivalent noise temperature, (a) The equivalent noise temperatures of the given four components are Waveguide 7 290(2-1) = 290K waveguide Mixer Tiger = 290(3- 1) = 580K Low-noise RF amplifier Tap = 290(1.7-1) = 203K Famplifier Typ = 290(5-1) = 160K (b) The effective noise temperature at the input to the LNA due to the antenna and waveguide is 441G, X Tantenna + (1 -G. ‘waveguide % Pantenna waveguide) * T waveguide = 0.2 50 + 290(1 - 0.2) = 42K The effective noise temperature of the system is, . T+ Te, + inixer Tip system = Te+Tpp + ; vt BE Gre” Gee Gainer 580 , 1160 2 eo 242-4203 + +5 526.2K %, = Power density at saturation G, = Gainof Im? BO, = Power back-off Figure of Merit Boltzmann constant in BK Losses due to rain = 105.0 dB-Hz 442where we have used the following gain of Im? antenna: G, = 10log 4xaxl ) (3x 108/14 x 10°)” = 44.5 dB Boltzmann constant k = -228.6 dB E, (©) Given the data rate in the uplink = 33.9 Mb/s and link margin of 6 dB, the required =? is 0 ei (Feet 5 (Tw — (10 log jy + 10log io) 105 - 6 - 10log19(33.9 x 106) " 105-6 -75.3 23.7 dB Equivalently, we have E, x = 234 0 Given the use of 8-PSK, the symbol error rate is defined by = ext [sina 0 For 8-PSK E b pete = 102 Ny 7 Wy 7 2x3 = 70 Hence, erfe(./702 x sin(m/8)) =0 443This result further confirms the statement we made in Example 8.2 in that the satellite communication system is essentially downlink-limited. Recognizing that we have more powerful resources available at an earth station then at a satellite, it would seem reasonable that the BER at the satellite can be made practically zero by transmitting enough signal power along the uplink. Problem 8.11 For the downlink, the relationship between ) (me) and ( Mn , expressed in decibels, is described by c) -() + 10lo; aS = (x gM + 10logR a where M is the margin and R is the bit rate in bits/second. Solving Eq. (1) for the the link margin in dB and evaluating it for the problem at hand, we get 10log gM = 85 ~10- 10log10(10°) = 5dB For the downlink budget, the equation for ( x) expressed in decibels, is as follows: (x, where k is Boltzmann's constant, ™ Gy = EIRP + (F ap ~ Ptexpace ~ 10108; gk downlink For a satisfactory reception at any situation, we consider additional losses due to rain etc. up to the calculated link margin of 5 dB. Hence, we may write G _ G (Fanaa 72? *()q Penn 18 ga t where EIRP = 57 dBW itee-space loss Lireespace 444= 92.4 + 2log ,9( 12.5) + 20log 19(40, 000) = 206 dB 10log jk = 228.6 dBK 10log gM = 5 dB Using these values in Eq. (2) and solving for G,/T, we get Gry (= 85-57 + 206 - 228.645 T Jas = 10.44B With 7: 310K, we thus find G, = 10.4 + 10log (310) = 35.31 dB ‘The receiving antenna gain in is given by 10log ) WO 10log 106, For a dish antenna (circular) with diameter D, the area A equals 2D?/4, Thus, 10log 9G, = 2Olog ,,D + 20log ,gf + 10log io(n) + 20.4(dB) where D is measured in meters and fis measured in GHz. Solving for the antenna diameter for the given system, we finally get Darin = 0.6 meters Problem 8.12 (@) Similarities between satellite and wireless communications: + They are both bandwidth-limited. + They both rely on multiple-access techniques for their operation. + They both have uplink and downlink data transmissions. + The performance of both systems is influenced by intersymbol interference and external interference signals. 445(b) Major differences between satellite and wireless communications: * Multipath fading and user mobility are characteristic features of wireless communications, which have no counterparts in satellite communications. + The cartier frequency for satellite communications is in the gigahertz range (Ku-band), whereas in satellite communications it is in the megahertz range. + Satellite communication systems provide broad area coverage, whereas wireless communications provide local coverage with provision for mobility in a cellular type of layout Problem 8.13, In a wireless communication system, transmit power is limited at the mobile unit, whereas no such limitation exists at the base station. A sensible design strategy is to make the path loss (.e., free-space loss) on the downlink as small as possible, which, in turn, suggests that we make (Path 1088)ypyink < (Path 1088) Joyntink Recognizing that path loss is inversely proportional to wavelength, it follows that Japlink > > downlink or, equivalently, Faptink 0.as as 0.25 5S “| 0 5 O25 —~ 0.95 ow Ons Oas— | 05 ' Ne Ors 0.25. oo S (0.105 0.125 | 1 S$, 0.125 0.105 0.125 Ce 5, O.1as ais 2 0.125 \ 5, d.obas & |o,ns 3 O.obs 466‘The Huffman code is therefore 8 10 Ss 11 8. 001 s3 010 8 011 8 0000 s 0001 ‘The average code-word length is 6 T= ¥ pele r= = 0.25(2)(2) + 0.125(8)(3) + 0.0625(4)(2) = 2.625 ‘The entropy of the source is 6 HS) = Dp be( 2) P=] Pk = 0.25(2) von sis + 0.125(3) vais i ) + 0.0625(2) veils wa = 2.625 ‘The efficiency of the code is therefore ‘We could have shown that the efficiency of the code is 100% by inspection since 4676 DY Px loga(/py) i ne 6 D Pele P=] where 1, = log,(/p,). Problem 9.13 a) ° S$ 07 ——~-07 s Os o 0.3 a] % on 1 ‘The Huffman code is therefore nn) 5; 10 s 11 The average code-word length is T = 0.7(1) + 0.15(2) + 0.12(2) =13 (b) For the extended source we have syatot [ve om [son [ose [oe [oer [ome [oo [om Probability | 0.49 | oo | eos | eos | o105 | o.o2n5 | o.25 | o.c22s | o-o225 $81 468Applying the Huffman algorithm to the extended source, we obtain the following source code: S080, S981. S982, 8189 8280 58181 Shy 8281, ae 1 oo1 010 011 0000 000100 000101 000110 000111 ‘The corresponding value of the average code-word length is Ty = 0.491) + 0.105(3)(3) + 0.105(4) + 0.0225(4)(4) Pry = 2,395 bits/extended symbol = 1.1975 bits/symbol (c) The original source has entropy According to Eq, (/0-28), HS) = 0.7 volar] + 0.15(2) wealats = 118 Hes) < © ¢H) + 2 a a This is a condition which the extended code satisfies.Problem 9.14 Symbol Huffman Code Code-word length A 1 1 B 011 3 c 010 3 D 001 3 E oo11 4 F 00001 5 G 00000 5 Problem 9.15 zt 0 00 2 ' 0 Vt a \ a ' OO op ; mal 0 4% \ Computer code Probability Huffman Code 00 1 0 2 11 1 10 4 o1 1 110 8 10 1 111 8 ‘The number of bits used for the instructions based on the computer code, in a probabilistic sense, is equal to 470obs ted. 1).2dits cpaneeea On the other hand, the number of bits used for instructions based on the Huffman code, is equal to ‘The percentage reduction in the number of bits used for instruction, realized by adopting the Huffman code, is therefore 100 x ag = 12.5% Problem 9.16 Initial step Subsequences stored: 0 Data to be parsed: 11101001100010110100.. Step 1 Subsequences stored: 0, 1, 11 Data to be parsed: 101001100010110100.. Step 2 Subsequences stored: 0, 1, 11, 10 Data to be parsed: 1001100010110100... Step 3 Subsequences stored: 0, 1, 11, 10, 100 Data to be parsed: 1100010110100... Step 4 Subsequences stored: 0, 1, 11, 10, 100, 110 Data to be parsed: 0010110100... 471Step 5. Subhsequences stored: 0, 1, 11, 10, 100, 110, 00 Data to be parsed: 10110100... Step 6 Subsequences stored: 0, 1, 11, 10, 100, 110, 00, 101 Data to be parsed: 10100... Step7 Subsequences stored: 0, 1, 11, 10, 100, 110, 00, 101, 1010 Data tobe parsed: 0 Now that we have gone as far as we can go with data parsing for the given sequence, we write Numerical 23 4 5 6 a 8 9 positions Subsequences 0, 1, 11, 10, 100, 110, 00, 101, 1010 Numerical 22, 21, 41, 31, 11, 42, 81 representations Binary encoded 0101, 0100, 0100, 0110, 0010, 1001, 10000 blocks 472Problem 9.17 48 o4, Plxy= plsys o z P(¥o) = (1 - p)p(xp) + p P(x) =(1- 1). yl -a-pG +m) 1 2 PCYy) = P P(X) + (1 - p) p(x) =p sa-pd = PG) + - pS) 1 2 473Problem 9.18 Px) = px) = ale ale =a-p) +p) Ply) = (1 = p)() + RS) 1 a who tee Py) wD a pn 3_P 4 2 Problem 9.19 From Eq.(9-52)we may express the mutual information as 1 4 PO} YK) 1¥) = ) Nogy| POF eM = BY, Mave a | ‘The joint probability p(x,y,) has the following four possible values: Oxy yy) = pO +B) where Po = (Xp) and py = P(x) The mutual information is therefore 474(-pp (-p) IQGY) = (1-1 ‘1-p) le pe aetna rola on] + CL-py) p logs (-pp p (=p) C-pp p * Pid-p) Pip + I TS pp oa ara Cp) al pi(l-p) + py(1-p) logy} —___—"__— ne rt + Ps Rearranging terms and expanding algorithms: IQGY) = p log, p + (1-p) log,(1-p) ~ [px(l-p) + (1-ppp] loge{p,(1-p) + (1-ppp] - [pw + (1p) (1-p)] loggfpyp + (1-pp) (1-p)] ‘Treating the transition probability p of the channel as a running parameter, we may carry out the following numerical evaluations: ‘=O: = Py logy py - (1 - py) logs (1 - py) .5, IOGY) = 1.0 IKY) = - 0.469 - (0.1 + 0.8 p,) logy (011 + 0.8 p,) - (0.9 - 0.8 py) logy (0.9 - 0.9 p,) P= 0.5, I(X;Y) = 0.531 p=0.2: IQGY) = - 0.7219 - (0.2 + 0.6 py) logy (0.2 + 0.6 py) = (0.8 - 0.6 p,) logs (0.8 - 0.6 p,) P) = 0.5, IQGY) = 0.278ICKY) = - 0.88129 - (0.8 + 0.4 p,) log, (0.3 + 0.4 p,) + (0.7 - 0.4 py) logy (0.7 - 0.4 py) P, = 0.5, 1(K:¥) = 0.1187 IY) =0 ‘Thus, treating the a priori probability p, as a variable and the transition probability p as a running parameter, we get the following plots: 1.0 Pz Problem From the plots of ICKY) versus p, for p as a running parameter, that were presented in the solution to Problem 10-19 we observe that I(X;Y) attains its maximum value at p,=05 for any p. Hence, evaluating the mutual information I(X:Y) at p,=0.5, we get the channel capacity: C = 1+ p logy p + (1 ~ p) logy (1 - p) - Hp) where H(p) is the entropy function of p. 476z = p(1 - p) + (1 - py) p = (1 - po) (1 - p) + pop Hence, Pip + (1 - py) =1-p, -p)-(1- ppp =1-2 Correspondingly, we may rewrite the expression for the mutual information I(X;Y) as IY) = H@) - H@) H(2) = ~ 2 logy 2 ~ (1 ~ 2) logy (1 - 2) H(D = - p logy p - (1 - p) logy (1 - p) (b) Differentiating I0GY) with respect to pp (or p,) and setting the result equal to zero, we find that 10G;¥) attains its maximum value at pp = p, = 1/2. (c) Setting po = p, - 1/2 in the expression for the variable z, we get: zel-z=12 Correspondingly, we have H(z) = 1 We thus get the following value for the channel capacity: C= IK) h, - p12 +1-H@) where H(p) is the entropy function of the channel transition probability p. 477Problem 9.22 From this diagram, we obtain (by inspection) Plyo Ix) = (1 - p)? + p? = 1 - 2p(1 - p) P(yo Ixy) = pCl - p) + (1 - pop = 2p - p) Hence, the cascade of two binary symmetric channels with a transition probability p is equivalent toa single binary symmetric channel with a transition probability equal to 2p(1 - p), as shown below: I= 2p(t-P) % 4 feo m) 2PCI-P) 4, * T=api-p) | Correspondingly, the channel capacity of the cascade is - Hp - p) = 2p(1 ~ p) loge (2p(1 - p)] - (1 - 2p + 2p”) loge(1 - 2p + 2p?) 478Problem 9.23 lea ° 00 a a 0 01 Lea = tog (PEED 1GY) = DY pe, wes (see) ° ‘The joint probabilities for the channel are P(%q Yo) = (1-1) pg P(X Yo) = 0 P(%q ¥2) = Pot P(xq¥,) = 0 P(x ¥1)) = (L- ep; P(X Y2) = Pye where po + p = 1. Substituting these values in (1), we get 1QGY) = (eof oles {-) + = pao {; “I Since the transition probability p = (1 - 0) is fixed, the mutual information I(X;Y) is maximized by choosing the a priori probability po to maximize H(po). This maximization occurs at po = 1/2,. for which H(p9) = 1. Hence, the channel capacity C of the erasure channel is I - ot 479Problem 9.24 332 (a) When each symbol is repeated three times, we have Messages Unused signals 000 001 010 11 100 101 110 iw ‘We note the following: The probal Ae Channel outputs 000 001 010 100 101 110 1 The probability that no errors occur in the transmission of three Os or three 1s is (1 - p)®. ity of just one error occurring is 3p(1 - p)”. ‘The probability of two errors ceeurringis 3p%(1 - p). The probability of receiving all three bits in error is p® With the decision-making based on a majority vote, it is clear that contributions 3 and 4 lead to the probability of error Ps = 3p%(1-p) + p? (b) When each symbol is transmitted five times, we have Messages Unused signals 00000 00001 00010 00011 11110 m1 Channel outputs 00000 00001 00010 00011 11110 mn faa‘The probability of zero, one, two, three, four, or five bit errors in transmission is as follows, respectively: (-p® 5p (1-py* 10p(1-p? 10p*(1-p? 5ptU-p) pe ‘The last three contributions constitute the probability of error P, = p> + 5p1-p) + 10p%1-p? (a) For the general case of n=2m + 1, we note that the decision-making process (based on a majority vote) makes an error when m+1 bits or more out of the n bits of a message are received in error. The probability of i message bits being received in error is 2h fap Hence, the probability of error is (in general) 7 P= Y (?}pa -pr tamer Li The results derived in parts (a) and (b) for m=1 and m=2 are special cases of this general formula. Problem 9.25 ‘The differential entropy of a random variable is independent of its mean. To evaluate the differential entropy of a Gaussian vector X, consisting of the components X,,X»,....X, We may simplify our task by setting the mean of X equal to zero. Under this condition, we may express the joint probability density function of the Gaussian vector X as 4812 \2 1 x x2 f(x) = 1 __oxp}-2_. - 2 (2K) 6469...0, 20; 208 ‘The logarithm of f(x) is loge f(x) = - logg((2n)"” 6409...) Hence, the differential entropy of X is W(X) = - Hele J fo ogg (fx(x)) dx = Togy((2n)"o102..0,) ff.» f fyoodx 2 + wwe fff [eB ‘We next note that Sf fixeddx = 1 SSS? tends = 0? iet.2,...0 Hence, we may simplify (1) as 482BR) = logg[(2n)”oy62...09] + Blogse = logy, [pxo2o? sel + F losse a lato’ 2m], 2 : 2 fantotod on ] 2 loge ~ 2 togaretoteg .. of!) When the individual variances are equal: Correspondingly, the differential entropy of X is (X) = z logy(2neo?) Problem 9.26 (a) The differential entropy of a random variable X is hO® = - if * f(x) logy fx(x)dx ‘The constraint on the value x of the random variable X is Ix Ism of Using the method! Lagrange multipliers, we find that h(X), subject to this constraint, is maximized when 483[2 [£400 loge £460 + 2 F400] is stationary. Differentiating -fy(x)logfy(x) + Af(x) with respect to f(x), and then setting the result equal to zero, we get logge + 2 = logy fy(x) This shows that f,(x) is independent of x. Hence, for the differential entropy h(X) under the constraints Ix |< M and gz fy(x)dx=1 to be maximum, the random variable X must be uniformly distributed: _{U2M, -Msx 0, where S,(f) is the power spectral density of the transmitted signal For the NEXT-dominated channel described in the question, the capacity is pp? log | 1+ PL Ja jie * eapolf) " va pr? lay v2 onl if } where B, k, J and f, are all constants pertaining to the transmission medium. This formula for capacity can only be evaluated numerically for prescribed values of these constants, 1 = 5 flog] i+ 2! z 493Problem 9.34 For k=1, Eq. (%/38) reduces to 10 logg(SNR) = 6 log,N + C, 4B @ Expressing Eq. (2.33) in decibels, we have 2 10 logg(SNR) = 6R + 10 logy [_2P (2) Max ‘The number of bits per sample R, is defined by R = loggN We thus see that Eqs. (1) and (2) are equivalent, with C, = 10 vent 494Problem 9.35 The rate distortion function and channel capacity theorem may be summed up diagrammatically as follows: mia OGY) max 20%) ota “tpansmissisn Dar compassion Limit Lit According to the rate distortion theory, the data compression limit set by minimizing the mutual information I(X;Y) lies at the extreme left of this representation; here, the symbol Y represents the data compressed form of the source symbol X. On the other hand, according to the channel capacity theorem the data transmission limit is defined by maximizing the mutual information I(X;Y) between the channel input X and channel output Y. This latter limit lies on the extreme right of the representation shown above. 495,1 Problem 9. 36 Matlab codes % Computer Problem in Chapter 9 % Figure: The minimum achievable BER as a function of 4% Eb/O for several different code rates using binary signaling. % This program calculates the Minimum required Eb/NO 4% tor BPSK signalling at unit power over AWGN channel 4% given a rate and an allowed BER 4% Code is based on Brandon’s C code. 4% Ref: Brendan J. Frey, Graphical models for machine % Learning and digital communications, The MIT Press % Mathini Sellathurai EbNo=double( (7.85168, 7.42122, 6.99319, 6.58785, 6.14714, 5.7329, 6.32711, 4.92926, 4.54106, 4.16588, 3.80312, 3.48317, 3.11902, 2.7981, 2.49337, 2.20617, . 1.93251, 1.67687, 1.43313, 1.20671, 0.994633, 0.794801, 0.608808, 0.434862, 0.273476, 0.123322, ~0.0148208, -0. 144486, -0.266247, -0.374365, -0.474747, -0.5708, 0.659038, ~0.736526, -0.812523, -0.878333, -0.944802, -0.996262, -1.04468, .. ~1,10054, 1.14925, -1.18536, -1.22298, -1.24746, -1.27304, -1.31061, -1.34588, -1.37178, 1.37904, -1.40388, -1.42563, -1.45221, -1.43447, -1.44302, -1.46129, 1.45001, -1.50485, -1.50654, -1.50192, -1.45507, -1.60877, 1.52716, -1.54448, “1.81713, 1.84378, ~1.5684]); rate= double([9.989372e-01, 9.980567e-01, 9.966180e-01, 9.945634e-01, 9.914587e-01, 9.868898e-01, 9.804353e-01, 9.722413e-01, 9.619767e-01, 9.490156e-01, 9,3346800-01, 9.185144e-01, 8.946454e-01, 8.715918e-01, 8.459731e-01, 8.178003e-01, 7.881055e-01, 7.565174e-01, 7.238745e-01, 6.900430e-01, 6.856226e-01, 6.211661 5.8664800-01, 5.526132e-01, 5.188620e-01, 4.860017e-01, 4.539652e-01, 4 3.938277e-01, 3.6533280-01, 3.382965e-01, 3.129488e-01, 2.889799e-01, 2.66187 1e-01, 2.4510790-01, 2.2516010-01, 2.068837e-01, 1.894274e-01, .. 4.73322Se-O1, 1.588691e-01, 1.453627e-01, 1.326278e-01, 1.210507e-01, 1.101604e-01, 1,002778e-01, 9.150450e-02, 8.347174e-02, 7.50800%e-02, 6.886473e-02, 6.266875e-02, 5 5 4 3 188306e-02, 4.675437e-02, 4.230723e-02, 3.8516370-02, 3.4760620-02, . 3.1852430-02, 2,8832460-02, 2.606097e-02, 2.3327900-02, 2.185325e-02, 1.981896e-02, 1.7641226-02, 1,6862210-02, 1.4441080-02, 1.314112¢-02]); N=68; bedouble([ie-6]); % Allowed BER 4 Rate R (bits per channel usage) redouble([1/32, 1/16,0.1,0.2,0.3,0.4,0.5, 0.6, 0.7, 0.8,0.85,0.95]); 496Lerzeros(t,length(r)); % initialize buffer for Eb/NO for pet:length(z) ¢ = doublo(r(p)#(1.0+b#10g(b)+(1.0-b) #10g(1.0-b) /1og(2.0))); sew; % Minimum Eb/WO calculations while ( (4-0) & (corabe(i)) ) isi-t; end isitty if ( (bo) | Gar) ) @ =double( EDNo(i)+(EbNo(i~1)-EbNo(i))#(c~rate(i))/(rate(i-1)-rate(i)) 1e(p)=10#10g10( (10° (0/10) }4c/e(p)); display(1e) else display('values out of range’) end end plot (10¥10g10(r) 16, *-") xlabel (Rate (4B)’) ylabel (Minimum E_b/_0 (4B)’) axis([10#20g10(1/32), 0, -2 4]) 497Computer Experiment in Chapter 9 Progran to create the figure for the minimum Eb/HO needed for error-free communication With @ rate R code, over an AVGN channel using binary signaling Thie program caleulates the Kinimun required Eb/NO for BPSK signalling at unit power over AVGN channel % given a rate and an alloed BER. % Code is based on Brandon’s C code. % Ref: Brendan J. Frey, Graphical models for machine % learning and digital communications, The MIT Press 4% Mathini Sellathurai Eblio= double([7.86168, 7.42122, 6.99319, 6.56785, 6.14714, 5.7329, 6.92711, 4.92026, 4.54108, 4.16568, 3.30312, 3.45317, 3.11902, 2.7981, 2.49337, 2.20817, 1.93251, 1.67687, 1.43313, 1.20671, 0.994633, 0.794801, 0.608808, 0.434862, 0.273476, 0.123322, -0.0148204, -0. 144486, -0.266247, -0.374365, -0.474747, -0.5708, 0.659038, -0.736526, -0.812523, -0.878333, ~0.944802, -0.996262, -1.04468, . “1.10084, 1.14925, -1. 18836, 1.22298, 1.24746, -1.27394, -1.31061, -1.34688, “1.37178, -1.37904, -1.40388, -1.42653, -1.45221, -1.49447, -1.44992, -1.46129, -1,45001, -1.60485, -1.50654, 1.60192, -1.45507, -1.60577, -1.52716, -1.54448, 1.61713, -1.84378, -1.5684)); x 9893720-01, 9.980567e-01, 9.9661808-01, 9.9456340-01, 9. 914587e-01, 9.8688980-01, 9.804953e-01, 9.7224130-01, 9.619767e-01, 9.490156e-01, 9.3346800-01, 9.1851440-01, 8.9464540-01, 8.715918e-01, 8.4597316-01, 8.178003e-01, 7, 881085e-01, 7.565174e-01, 7.238745e-01, 6.900430e-01, 6.556226e-01, 6.2116616-01, 5.8664800-01, 5.525132e-01, 5.188620e-01, 4.860017e-01, 4.5396520-01, 4. “01, 3.938277e-01, 3.653328e-01, 3.382965e-01, 3.129488e-01, 2.889799e-01, 2 “01, 2.451079e-01, 2.251691e-01, 2.068837e-01, 1.8942740-01, .. a “01, 1.8885910-01, 1.453627e-01, 1.326278e-01, 1.210507e-01, 1.101604e-01, 1 “01, 9.1604600-02, 8.3471746-02, 7.698009e-02, 6.886473e-02, 6.266875e-02, 5.6988470-02, §.188306e-02, 4.675437e-02, 4.2397230-02, 3.851637e-02, 3.4760620-02,.. 3.1852430-02, 2.8832460-02, 2.606097e-02, 2.332790e-02, 2.185326¢-02, 4,941896e-02, 1.7641226-02, 1.5862210-02, 1.4441080-02, 1.3141128-02]); 8; bedouble(0.5:~19-6:16-6); % Allowed BER jouble( [0.99,1/2,1/3,1/4,1/5,1/8]); % Rate R(bits/channel usage) Ae=zeros(t,1ength(b)); Length») double (r# (1 .0+0(p)*10g(b(p) )+(41.0-b(p) )#Log(4.0-b(p))/20g(2.0))); 498,acu; while ( (i>=0) & (erate(i)) ) isi-t; end ait if ( (0) | Gav) ) fe = double(EbNo(i)+(EbNo( 1-1) -EbNo(i))#(c-rate(i))/(rate(i-1)-rate(i)) 1e(p)=10¥10g10((10"(e/10) }#c/r) 5 else display(*values out of range’) end end plot (ie, 10¥10g10(b),-") end xlabel(’E_b/N_O (4B)?) ylabel (Minimum BER?) axis([-2 1-50 -10]) 499Answer to Problem 9.36 Minimum EN (8) ° Rate (8) Figure 1: ‘The minimum Eb/INO needed for error-free communication with a rate R code, over an AWGN channel using binary signalingee oe -15| -25| & 1 a 13, § -s0] & = -3s| +0] 45} a5 7 -05 @ os 1 5M, (68) Figure 2: ‘The minimum achievable BER as a function of Eb/NO for several different code rates using binary signaling, 501Chapter 10 Error-Control Prot ‘The matrix of transition probabilities of a discrete memoryless channel with 2 inputs and Q outputs may be written as fe b) pt. b) pb) ~ p@-1b) 0h) path) ph) ~ pQ-1 hy, For a symmetric channel, pGb) = pQ-1-jl), j= 0,1,...Q-1 Moreover, each row of the matrix P contains the same set of numbers, and each column of the matrix P contains the same set of numbers. For example, for Q=4, we may write pe ft abb Ib ba a The sum of the elements of each row of matrix P must add up to one. Hence, for this example, 2a +2b=1 The probability of receiving symbol j is PG) = pG b) pO) + pG pay For equally likely input symbols: 502348 =pay=2 p(0) = pil) a Hence, PG) = 2 [pd b) + pQ-1 - 5b] For the example of Q=4, we have PO=l@+b ei j= 0,123 In general, we may write FOL Q-1 p=, PG a Problem 10.2 For a binary PSK channel, the probability density function of the correlator output in the receiver is fxx by = —b on 1 + fa | No f(x hy = 1 onl ~~ al | Lety= |2 x 0 2 =|2 a dy No y pertains to a Gaussian variable of mean + express the channel transition probability as 25 py b) = Li exp] - A ]y + |= V2n 2 No 2E py lt) = LE exp] - F]y - |=* 2x 2 No where - 0 | Fir peas hep adap Given that G=P ih] H=[[, 1 P?| we find that the parity-check matrix is 100:1110 H-l010:0111 oor:i10L In Example 3 of the text, the message sequence (1001) was applied to the encoder and the output code word was 0111001. For the above encoder, the parity bits are 110 and the code word is then 1101001. In particular, we have 513Shift Input Register contents 000 1 1 101 2 0 111 3 0 110 4 1 110 If we were to make an error in the middle bit and receive 1100001, then circulating it through the syndrome calculator, we have Shift Input Register contents 000 1 1 100 2 0 010 3 0 001 4 0 101 5 0 qi 6 1 010 1 1 101 From the parity check matrix we see that the syndrome calculator output 101 corresponds to the error pattern 0001000. The corrected code word is therefore 1101001. Problem 10.11 ‘The error polynomial is e(X) = (X) + eX) We are given sideX) =X +X? +X3+x6 rX) = X+X3+x6 ‘The error polynomial is therefore e(X) = xX? Consider next the syndrome polynomial s(X). The syndrome calculator for the generator polynomial eX) =1+X+x3 is shown in Fig. 10.11; this calculator is reproduced here for convenience of presentation: _ 10100) Flip- ——Mod-2 flop adder Circulating the received bits through the syndrome calculator, we may construct the following table: Tnitial state 000 100 010 oo1 010 001 010 001 515Received bit Here, the syndrome polynomial is s(X) = X? which, for the problem at hand, is the same as the error polynomial. This result demonstrates the property of the syndrome polynomial, stating that it is the same as the error polynomial when the transmission errors are confined to the parity-check bits. In Problem 11.11 the third parity-check bit is received in error. Problem 10.12 The encoder structure is Fle Pep Mya The syndrome calculator is 516Problem 10.13 (@) A maximal-length code is the dual of the corresponding Hamming code. The generator polynomial of a.(15,11) Hamming code is given as | + X + X4, We may therefore define the feedback connections of the corresponding (15,4) maximal-length code by choosing the primitive polynomial W(X) = 14X4Xx" ‘The feedback connections are therefore [4,1], which agrees with entry 3 of Table 7.1 Specifically, the encoder of the (15,4) maximal-length encoder is as follows: Téa AY aia : ot Arte (b) The generator polynomial of the (15,4) maximal-length code is tex exh WX) taxa? B(X) = Performing this division modulo-2, we obtain B(X) = 14X44 Xa eX exe x! (This computation is left as aj exercise for the reader.) Assuming that the initial state of the encoder is 0001, we find that the output sequence is (111101011001000). Here we recognize that the length of the coded sequence is 2+ - 1 = 15. The output sequence repeats itself periodically every 15 bits. 517Problem 10,14 (a) n=2"-1=31 symbols Hence, the number of bits per symbol in the code is m = 5 bits (b) Block length = 31 x 5 = 155 bits (©) Minimum distance of the code is dyin = 2t+1 =n-k+1 =31-15+1 = 17 symbols (d) | Number of correctable symbols is t=in-w = 8 symbols 518Problem 10,15 ‘The encoder is realized by inspection: 2 = (1,0,1) 2 = (1,1,0) ge = (1) For the Hamming code, the parity check matrix H is more structured than that for the repetition code. Indeed, the matrix H for the Hamming code includes that for the repetition code as a submatrix. Problem 10.16 Input Flup- fe0 toble Using this encoder, we may construct the following!by inspection: Message 1 O 1 1 1 1 Output 11 610)=«11 “O1 o1 01 Qo Original message The code is in fact systematic. 519The generator polynomials are eM =1+xX+x?+x3 gH =1+X+ x3 The message polynomial is m(X) = 1+X?+X3+X*+.., Hence, eK) = gay moxy =1+X+X34X4+ X54 © X) = gx) mx) =1+X+Xx?+x8+ Xo4X74 Henee, 520fe) = 1,1,0,1,1,1,... {ce} = 1,1,1,1,0, The encoder output is therefore 11, 11, 01, 11, 10, 10. Problem 10.18 The encoder of Fig. (#13 (b) has three generator sequences for each of the two input paths; they are as follows (from top to bottom) e-an g®-a0, g-an 9-00, =n, 2 = 00) Hence, ePO-1+%, gPO=1, e009 =1+X Bs =X ey =1+xX, ga =0 The incoming message sequence 10111... enters the encoder two bits at a time; hence mY 211... m® =01 The message polynomials are therefore m(X) = 14+X+4.. m,(X) =X + Hence, the output polynomials are 521Max) = g(PaK) m0) + g§fPCK) meCK) =(1+X)(1+X%+..)+XK +.) OEE pn Qk) = g)70q) my) + gCK) mK) = 1+X+.) +1 +X) XK +. =1+X+ +X+X? 4. =1+X? +. cx) = gi0x) m+ gC) mK) =O OMX ++ OK +.) =1+X? The output sequences are correspondingly as follows: 21,0... = 1,0, = 1,0,.. The encoder output is therefore (1,1,1), (0,0,0), ... Problem 10.19 1. If the input sequence is 00, the encoder output: is 00, 00, 00, 00. 2, If the input sequence is 11, the message polynomial is m(X)=1+X ‘The two generator polynomials areeg =1+X+x? gx) =1+x? Hence, DK = +X) +X + x4) =1+Xx3 0X) = (1 +X) (1 + X%) =1+X+x?2+x8 The encoder output is 11, 01, 01, 11 3, If the input sequence is 01, the message polynomial is m(X) = X Hence, eX) = X(1 +X + X%) =X+X?+x3 c@X) = X(1 + X) =X+x3 The encoder output is 00, 11, 10, 11 4. Finally, if the input sequence is 10, the message polynomial is m(&X) =1 Correspondingly, 523ay) = gan =1+X+x? 0 2x) on gx) =1+x? Hence, the encoder output is 11, 10, 11, 00. The encoder outputs calculated above are in perfect accord with the entries of the code tree 00.00 00 524Problem 10.20 525 The output sequence is 11, 10, 11, 01, 01Problem 10.21. 526Problem 10.22 The encoder of Fig. Pi0/Thas eight states: Ico tate Register contents 000 100 010 110 oo1 101 oll 111 rRae aoe The state diagram of the encoder is as follows: In this diagram, a solid line corresponds to an input of 0 and adashed line corresponds to an input of 1. 527Problem 10.23 (a) The encoder of Fig. 10/3} has four states: State Register contents a 0,0 b 1,0 c 0,1 d 14l In the state diagram shown below, each branch is labeled with the input dibit followed by the output triplet. war tofarn 00/000 (b) Starting from the all-zero state a, the incoming sequence 10111... produces the path a d... Equivalently, we have the decoded (output) sequence (111), (000), ..., which is exactly the same result calculated in Problem 10.18. <5.Problem 10.24 An MSK system has two distinct phase states State Phase,radians a 0 b r The transmission of a 1 increases the phase by 1/2, whereas the transmission of a 0 decreases the phase by 1/2. Correspondingly, the transmission of dibit 10 or 01 leaves the state of MSK unchanged, whereas the transmission of dibit 00 or 11 movesthe system from one state to the other. For the output, we have Input dibit Qutput frequencies ae ff 01 f fy 10 f, 00 fy & tw'Ae, The trellis diagram for MSK is as follows:Since d,,;, = 5 and the number of errors in the received sequence is 2, it should be possible to decode the correct sequence. This is readily demonstrated by applying the Viterbi algorithm. Problem 10.25 Received 00 bits 10 00 10 00 00 Notations @ patR metric cay Banh metric — message bit 6 aes Message bit I Ke debate pat 5 uke above figure we See that fhe daccetedd SLom eo bag for sequence is 00000000 0000--, themby coppsel i the two errors the Aecaived Sequers + 530Problem 10.26 (a) Coding gain for binary symmetric channel is 10 x 1/2 2 Gq = 10 logy 4 = 10 logy 2.5 =4eB (b) Coding gain for additive white Gaussian noise channel is Gq = 10 logio (» x = 10 logy 5 =74B em 1 The trellis of Fig, P/o-27 corresponds to binary data transmitted through a dispersive channel, viewed as a finite-state (i.e., two-state) machine. There are two states representing the two possible values of the previous channel bit. Each possible path through the trellis diagram of Fig. Pio-27 corresponds to a particular data sequence transmitted through the channel. To proceed with the application of the Viterbi algorithm to the problem at hand, we first note that there are two paths of length 1 through the trellis; their squared Euclidean distances are as follows: 42, = (0 - 1? = 001 20 -(- 2 = af, = 2.0 -(-.9)? = 861 5,Each of these two paths is extended in two ways to form four paths of length 2; their squared Euclidean distances from the received sequence are as follows: (@) 2, = 0.01 + (0.0 - 1.1? = 1.22 2, = 3.61+ (0.0 - 0.9) = 4.42 ) 42, = 0.01 + (0.0 - (-0.9)? = 0.82 42, = 3.61 + (0.0 - (1.0)? = 4.82 Of these four possible paths, the first and third ones (j.e., those corresponding to squared Euclidean distances d”, , and d’, 5) are selected as the "survivors", which are found to be in agreement. Accordingly, a decision is made that the demodulated symbol ap=1. Next, each of the two surviving paths of length 2 is extended in two ways to form four new paths of length 3. The squared Euclidean distances of these four paths from the received sequence are as follows: (a) a3, = 1.22 + (0.2 - 1.1? = 2.03 dj'y = 0.82 + (0.2 - 0.9? = 131 ) jig = 1.22 + (0.2 - (-0.9)? = 2.43 a3, = 0.82 + (02 - (-1.) = 2.51 532This time, the second and third paths (i.e., those corresponding to the squared Euclidean distances d’3,2 and d,.,) are selected as the "survivors". However, no decision can be made on the demodulated symbol a, as the two paths do not agree. To proceed further, the two surviving paths are extended to form two paths of length 4. The squared Euclidean distances of these surviving paths are as follows: @ a2, = 1.31 + (-1.1 - 11? = 6.15 dz, = 2.43 + (-1.1 - 0.97 = 6.43 ) fy = 1.81 + (-1.1 - (-0.9)? = 1.35 4}, = 243 + (-1.1 - (-1.1)? = 2.43 ‘The first and third paths are therefore selected as the "survivors", which are now found to agree in their first three branches. Accordingly, it is decided that the demodulated symbols are ay = +1, a, = -1, and ag = +1. It is of interest to note that although we could not form a decision on a, after the third iteration of the Viterbi algorithm, we are able to do so after the fourth iteration. Figure 1 shows, for the problem at hand, how the trellis diagram is pruned as the application of the Viterbi algorithm progresses through the trellis of Fig. P11. 533Lteration b Tteration( Tterahon2 Tterebiea 3 eos 0.01 1.22 131 ous 3.41 0.82 243 rae Demodal Gs ee oe @) ane Fig. (This problem is taken from R.E. Blahut, "Digital Transmission of Information", Addison- Wesley, 1990, pp. 144-149.The interested reader may consult this book for a more detailed treatment of the subject.) 534a fh. fyProblem 10.29 (a) Without coding, the required E/N is 12.5 dB. Given a coding gain of 5.1 dB, the required EyJNo is reduced to = 74 dB For the downlink, the equation for C/Ng is rc G = = EIRP + mia” 8? 47 (b) By definition, the formula for receive antenna gain is k free-space * 4nA, we where A, is the receive antenna aperture and A is the wavelength. Let (A, coding denote the receive antenna aperture that results from the use of coding. Hence 4nd, (4M AL) coain 10g, ~ 1010g,{ — "= or, equivalently, love cod 5.1 dB Hence, “+ = antilog 0.51 = 3.24 @), a coding The antenna aperture is therefore reduced by a factor of 3.24 through the use of coding. Expressing this result in terms of the antenna dish diameter, d,we may write 5362 ada ( 3.24 R(deoding) which yields Diameter_of antenna without coding _d_ 2 R54 = 1 Diameter of antenna with coding —~ dooging ‘That is, the antenna diameter is reduced by a factor of 1.8 through the use of coding, Problem 10.30 Nonlinearity of the encoder in Fig. P10,30 is determined by adding (modulo-2) in a bit-by-bit manner a pair of sets of values of the five input bits (1p. Layyet> Ziyn2+ ay Mont} and the associated pair of sets of values of the three output bits Yo,, ¥1,, and ¥2 ,.. If the result of adding these two sets of values of input bits, when it is treated as a new set of values of output-bits, does not always give a set of values of input bits identical to the result of adding the two sets of values of the aforementioned output bits, then the convolutional encoder is said to be nonlinear, For example, consider two sets of values for the sequence {24 5 4 n-1s fyn-2» Lyn lant }s that are given by {0,0,1,1,1} and {0,1,0,0,0}. The associated sets of values of the three output bits Fos Yiye ¥y are {0,1,1) and {1,0,0}, respectively. If the 5-bit sets are passed through the Exclusive OR (ice., mod-2 adder) bit-by-bit, the result is {0,1,1,1,1). If the resulting set (0,1,1,1,1} is input into the encoder, then the associated output bits are {1,1,0}. However, when the sets of output bits (0,1,1} and {1,0,0} are passed through the Exclusive OR, bit-by-bit, the result is {1,1,1}. Since the two results {1,1,0) and (1,1,1) are different, it follows that the convolutional encoder of Fig, P10.30 is nonlinear. Problem 10.31 Let the code rate of turbo code be R. We can write (a 537= Uth-P Hence R= p/(q,+%-P) Problem 10.32 Figure 1 is a reproduction of the 8-state RSC encoder of Figure 10.26 used as encoder 1 and encoder 2 in the turbo encoder of Fig. 10.25 of the textbook. For an input sequence consisting of symbol | followed by an infinite number of symbols 0, the outputs of the RSC encoders will contain an infinite number of ones as shown in Table 1 Fig. 1 b= a@c@e b@c@dGe Initial conditions: ¢ = d= e = 0 {empty} input Intermediate mpurs foutpur] a PE © ad € f T T 0 0 0 T 0 1 I 0 0 0 0 1 1 1 0 It 0 ) 1 1 1 1 0 1 0 1 1 1 0 0 1 0 1 0 0 0 0 1 0 A 0 1 0 0 1 0 0 1 1 0 0 0 The output is 1011101001110100111 538Therefore, an all zero sequence with a single bit error (1) will cause an infinite number of channel errors. [Note: The all zero input sequence produces an all zero output sequence.] Problem 10.33, (a) 4-state encoder x * (systematic bits) Sol d Parity check bits z 8-state encoder _ * (systematic bits) Parity check bits z 16-state encoder XO * X (ey ctematic bits) Parity check bits 539{b) 4-state encoder 1+D+D "14D? a(D) = [: By definition, we have (2m) _1i+D+D° \M(D) 1+D where B(D) denotes the transform of the parity sequence {b;} and M(D) denotes the transform of the message sequence {m;}. Hence, (1+D°)B(D) = (1+ D+ D")M(D) ‘The parity-check equation is given by (mm; + m,.. +m;.2) + (b+ bj) = 0 where the addition is modulo-2. Similarly for the 8-state encoder, we find that the parity-check equation is, m+ m9 +m,4 4b) + bj) + big tis = 0 For the 16-state encoder, the parity-check equation is im, + m4 +B, +B, + Bint Bi + Bpg = 540Problem 10,34 (a) Encoder oi +[ENCy > oy Ly, tp, +, Oty, are M interleavers ENC), ENC3, «++, ENCy are M recursive systematic convolutional (RSC) encoders Z isthe message sequence 21s 2qp--% Zyy are the resulting M parity sequences (b) Decoder L, {L(n},i#l a -$ Ly(n+1) -[@ }-—+paq, a - (Lio) #2 | D - 1 > >| a {DEC >| A a ) * ey a -X_ Ly(ntl) > L, OM *fpECy Tay 4 s 0 Dlock_errors(iteration) = block_errors(iteration) +1; end end Yotal bit errors total_errors=total_errorst bit_errors; 4% bit error rate 544if block_errors(no_of_iterations)==block_error_limit BER(i:noof_iterations)= total_errors(1:no_of_iterations)/ block_number/(block_size-m); end end 545function output = turbo_encorder( Data, code_g, Alpha) % Turbo code encorder ", Used in Problem 10.36, CS: Haykin WM. Sellathurai [n,K] = size(code_g); mik-4; block_s = length(Data); state = zeros(m,1); sros(3,block_s#n) ; % encorder 1 for isi: it i ak elseif i> blocks dk = rem( code_g(1,2:K)astate, 2); end ak = ren( code_g(1, vk = code_g(2, 1)#a1 [dk ;state], 2); for j = 2K vik = xor(v_k, code_g(2,j)+state(j-1)); end; state = [a_k;state(1:m-1)]; y(1,i)=4_k; y(2,i)=v_x; end Yencorder 2 4% interleaving the data for i= 1: block.stm yeilde(1,i) = y(1,atpha(i)); end state = zeros(m,1); 4% encorder 2 for i = 1: block_stm dk = ybilde(s,i); ak = ren( codeg(1,:)#[4k ;statel, 2); vk = code_g(2,1)*a_k; for j = 2:K vk = xor(v_k, code_g(2,j)#state(j-1)); endstate(t:m-1)1; % inserting odd and even parities for ist: block_s¢m output (1,nei-i) = 24y(1,4)-1; if rem(i,2) output(1,n6i) = 24y(2,4)=15 else output(i,n¢i) = 2¥y(3,4)-1; end end S47function [axt_o, nxt. Yused in Problent0.36. 4% code trellis for RSC; % Mathini Sellathurai Ist_o, Ist_s] = enc_trellis(code_g); % code properties [n,K] = size(code_g); m=K~ 4; no_of_states = 2°m; for s=1: no_of_states dec_ent_s=8: mis det % decimal to binary state while dec_ent_s >=0 & icom bin_ent_s(i) = rem( decent_s,2) ; dec_ent_s = (dec_ent_s~ bin_ent_s(i))/2; +15 end bin_ent_s=bin_ent_s(m:-1:1); next state when input is 0 dk = 0; ak = rem( code_g(t. vk = code_g(2,1)#a_k {0 binent_s 1’, 2); for j= 1:Ke1 vik = xor(v_k, code_g(2,j+1)*bin_ent_e(j)); end; nstateO = [ak bin_ent_e(1:m-i)]; y-0 = [0 vk]; Y% next state when input is 1 dk =; ak = rem( code_g(t,:)#L1 binent.e]’, 2); vik = code_g(2,1)#a_k; for j= 1:K-1 vk = xor(v_k, code_g(2, j+1)#bin_ent_s(j)); end; astatei = [a_k binent_s(1:m-1)]; yt vad; next output when input 0 1 nxt_o(s,:) = [y_0 y_t]; 5484% binary to decimal state (m-1:-1:0); dstateO=nstateded’+1; dstatei=nstateitd’+1; % next state when input 01 axt_s(s,:) = [dstated dstatei J; 4% finding the possible previous state frm the trellis Ist_s(nxt_s(s, 1), 1) Ast_s(axt_s(s, 2), 2)=5; Ast_o(nxt_s(s, 1), 1:4) = nxto(e, 1:4) ; Ast_o(axt_s(s, 2), 1:4) = nrto(s, 1:4) } ond 549function output = demultiplex(Data, Alpha); 4% demultiplexing the received signal % used in problem 10.36, CS: Haykin % Mathini Sellathurai Dlock_s = fix(Length(Data)/2); output=zeros(2,block_s); for i= t: blocks Dataf(i) = Data(2+i-1); if ren(i,2)>0 output (1,26i) = Data(2ri); else output (2,241) end Data(2¥i); end for i = t:block_s output (1,2#1-1) output (2, 244-1) end Dataf(i); Dataf(Alpha(i)); 8function L = BCJLi(Datar, code_g ,apriori) 4% Log-BCJL (LOG-MAP algorithm) for decoder 1 4% Used in Problem 10.36, CS: Haykin % states, memory, constraint length and block size Dlock_s = fix(Length(Datar)/2); [n,K] = size(code_g); m=K~4; no_of_states = 2"m; ingty = 1610; zero=16-300; 4% forward recursion alpha(1,1) alpha( 1,2: = cinftytones(1,no_of_states-1); % code-trellis [nxt_o, nxt_s, 1st_o, 1st_s] = cnc_trellis(code_g); axt_o = 2enxt_o- Ast_o = 2+lst_o-t; for i= t:block.s for ent. branch = -inftysones(1,no_of_etates); branch(1st_s(cnt_s,1)) = -Datar(2+i-1)+Datar(2ei)# Lst_o(ent_s,2)~log(1+exp(apriori(i))); branch(1st_s(cnt_s,2)) = Datar(2ei-1)+Datar(2#i)*.. Lst_o(cnt_s,4)+apriori (i)~Log(1+exp(apriori(i))); it (sum(exp(branchtalpha(i,:)))>zero) alpha(it1,cnt_s) = log( sum( exp( branchtalpha(i,:)))); else alpha(i+t,cnt_s) =-t#infty; end end alpha_max(iti) = max(alpha(i+t,:)); alpha(iti alpha(i+i,:) - alpha_max(itt); end 4% backward recursion beta(block_s, 1) beta(block_s,2:no_of_states) = ~inftytones(1,no_of_states-1); for i = block s-1:-1:1 for ent_s = 1:no_of_states 551branch = -inftytones(1,no_of states); branch(nxt_s(cnt_s,1)) = “Datar(2eiti)+Datar(2#i+2)*.. axt_o(ent_s,2)~1og(1¥exp(apriori(i+1))) ; branch(axt_e(cnt_s,2)) = Datar(2+iti)+Datar(2#i42)*... axt_o(ent_s,4)+apriori (i+1)-Log(i+exp(apriori(i#1))); it (sun(exp(branchtbeta(i+1,:)))zero) alpha(it1,cnt_s) = 1og( sum( exp( branchtaipha(i,:)))); else alpha(i+t,cnt_s) =-1eintty; end end alpha_max(itt) = max(alpha(i#t,:)); alpha(i+t,:) = alpha(iti,:) - alpha_max(itt); end % backward recursion beta(block_s,1)=0; beta(block_s,2:no_of_states) = ~inftyeones(1,no_of_states-1); for i = block_s-t:-4:1 for cnt_s = 1:no_of_statesbranch = -inftytones(1,no_of_states) ; branch(nxt_s(cnt_s,1)) = -Datar(2+i+i)+Datar(2*i#2)s. axt_o(cnt_s,2)~1og(1+exp(apriori (it1))); branch(nxt_s(cnt_s,2)) = Datar(2+it1)4Datar(2ei+2)*... axt_o(ent_s,4)+apriori(itt)-1og (1+exp(apriori(i#1))); it (sum(exp(branch+beta( itt, :)))

You might also like

• Advances in Microwaves: Volume 3

  

  From Everand

  Advances in Microwaves: Volume 3

  Leo Young

  No ratings yet
• BER For BPSK in Rayleigh Channel

  

  Document6 pages

  BER For BPSK in Rayleigh Channel

  Bùi Thế Anh

  No ratings yet
• Mathematical Theory of Connecting Networks and Telephone Traffic

  

  From Everand

  Mathematical Theory of Connecting Networks and Telephone Traffic

  Elsevier Books Reference

  No ratings yet
• • Magazines
  • Podcasts
  • Sheet music
• Unit 3

  

  Document6 pages

  Unit 3

  ShanilDayalan

  No ratings yet
• Antenna Theory and Applications

  

  From Everand

  Antenna Theory and Applications

  Hubregt J. Visser

  Rating: 5 out of 5 stars

  5/5 (1)
• Communication Principles Chapter II - Amplitude Modulation

  

  Document87 pages

  Communication Principles Chapter II - Amplitude Modulation

  marwan

  No ratings yet
• Top-Down Digital VLSI Design: From Architectures to Gate-Level Circuits and FPGAs

  

  From Everand

  Top-Down Digital VLSI Design: From Architectures to Gate-Level Circuits and FPGAs

  Hubert Kaeslin

  No ratings yet
• Solved Problems

  

  Document10 pages

  Solved Problems

  Ravindra Kumar

  No ratings yet
• Electrical Overstress (EOS): Devices, Circuits and Systems

  

  From Everand

  Electrical Overstress (EOS): Devices, Circuits and Systems

  Steven H. Voldman

  No ratings yet
• AM Problems

  

  Document13 pages

  AM Problems

  pad pest

  No ratings yet
• Nonlinear System Analysis

  

  From Everand

  Nonlinear System Analysis

  Austin Blaquiere

  No ratings yet
• Ece V Analog Communication (10ec53) Notes

  

  Document74 pages

  Ece V Analog Communication (10ec53) Notes

  Santhosh Kumar H O

  No ratings yet
• Computer Techniques for Electromagnetics: International Series of Monographs in Electrical Engineering

  

  From Everand

  Computer Techniques for Electromagnetics: International Series of Monographs in Electrical Engineering

  R. Mittra

  Rating: 3 out of 5 stars

  3/5 (1)
• Electromagnetic Waves and Transmission Lines

  

  Document180 pages

  Electromagnetic Waves and Transmission Lines

  Gurusreenu

  No ratings yet
• Fundamentals of Electronics 2: Continuous-time Signals and Systems

  

  From Everand

  Fundamentals of Electronics 2: Continuous-time Signals and Systems

  Pierre Muret

  No ratings yet
• Assignment Communication Engineering: Lavish Bagga 18105128

  

  Document17 pages

  Assignment Communication Engineering: Lavish Bagga 18105128

  Anonymous Xw1jpCpV5g

  No ratings yet
• Spacecraft Attitude Control: A Linear Matrix Inequality Approach

  

  From Everand

  Spacecraft Attitude Control: A Linear Matrix Inequality Approach

  Chuang Liu

  No ratings yet
• Ece 306 - HW 1

  

  Document1 page

  Ece 306 - HW 1

  sightlesswarrior

  No ratings yet
• Analog Modulation Techniques

  

  Document66 pages

  Analog Modulation Techniques

  pawan

  33% (3)
• LICA Lecture Notes by A.mounika

  

  Document77 pages

  LICA Lecture Notes by A.mounika

  Anonymous 4u5XkWGO

  No ratings yet
• Amplifiers and Oscillators

  

  Document10 pages

  Amplifiers and Oscillators

  Kapil Choudhary

  No ratings yet
• Design of LiDAR System

  

  Document9 pages

  Design of LiDAR System

  dipak1304

  No ratings yet
• Solution Manual Antenna Theory by Balanis Edition2 Chapter15b 3036

  

  Document20 pages

  Solution Manual Antenna Theory by Balanis Edition2 Chapter15b 3036

  Subrata Maiti

  No ratings yet
• Antenna Theory Analysis and Design - (Chapter 2 Fundamental Parameters and Figures-Of-Merit of Antennas)

  

  Document102 pages

  Antenna Theory Analysis and Design - (Chapter 2 Fundamental Parameters and Figures-Of-Merit of Antennas)

  andy fiutur

  No ratings yet
• EC8453 OPAMP Applications Guide

  

  Document6 pages

  EC8453 OPAMP Applications Guide

  ShanilDayalan

  No ratings yet
• Am Solved Problems

  

  Document10 pages

  Am Solved Problems

  sukhi_digra

  100% (2)
• Stray Current Protection in DC Rail Systems

  

  Document10 pages

  Stray Current Protection in DC Rail Systems

  faarig

  No ratings yet
• Signals and Systems

  

  Document3 pages

  Signals and Systems

  Arjun Arjun

  100% (1)
• Solutions Manual For: Arm Assembly Language

  

  Document129 pages

  Solutions Manual For: Arm Assembly Language

  mohammed Irheem

  No ratings yet
• 15EC306J Cycle 2 Expts

  

  Document28 pages

  15EC306J Cycle 2 Expts

  Sanjeev Kumar

  No ratings yet
• Signals and Systems

  

  Document30 pages

  Signals and Systems

  Mohammad Gulam Ahamad

  100% (3)
• Linear Integrated Circuits Course Outline PDF

  

  Document49 pages

  Linear Integrated Circuits Course Outline PDF

  Sarvjeet Singh Sohal

  No ratings yet
• LabSheet Experiments - Giatutinhyeu14 - 2

  

  Document176 pages

  LabSheet Experiments - Giatutinhyeu14 - 2

  Nhi Ot

  No ratings yet
• Ec 6511 Digital Signal Processing Lab Manual PDF

  

  Document88 pages

  Ec 6511 Digital Signal Processing Lab Manual PDF

  Bala913

  No ratings yet
• EMF UNIT V Waveguides

  

  Document98 pages

  EMF UNIT V Waveguides

  sravanti kanugu

  No ratings yet
• Linear Technology-Understanding IP2 and IP3 Issues in Direct Conversion Receivers For WCDMA Wide Area Basestations

  

  Document5 pages

  Linear Technology-Understanding IP2 and IP3 Issues in Direct Conversion Receivers For WCDMA Wide Area Basestations

  Michael Graham

  No ratings yet
• UNIT I, II, III, IV, V Notes PDF

  

  Document93 pages

  UNIT I, II, III, IV, V Notes PDF

  Thiagu Rajiv

  No ratings yet
• Capacitors and Inductors

  

  Document68 pages

  Capacitors and Inductors

  Jhasper Managyo

  No ratings yet
• Electronic Devices - Circuits Lab Manual

  

  Document77 pages

  Electronic Devices - Circuits Lab Manual

  pdnkiran

  No ratings yet
• DPSK Modulation Explained

  

  Document4 pages

  DPSK Modulation Explained

  Mochammad Andi Afriansyah

  100% (1)
• Control System Question Bank

  

  Document3 pages

  Control System Question Bank

  manish_iitr

  No ratings yet
• Biasing Circuits for Transistors Explained

  

  Document65 pages

  Biasing Circuits for Transistors Explained

  sai inesh

  No ratings yet
• Fourier Transform

  

  Document132 pages

  Fourier Transform

  Asif Noor

  100% (1)
• MIT 6.341 Lecture 20 - The Goertzel Algorithm and Chirp Transform

  

  Document4 pages

  MIT 6.341 Lecture 20 - The Goertzel Algorithm and Chirp Transform

  c_mc2

  No ratings yet
• Electronic Communication Trainer

  

  Document1 page

  Electronic Communication Trainer

  Fandeto Gaming

  100% (1)
• 08-Com101 AM

  

  Document11 pages

  08-Com101 AM

  Hồng Hoan

  No ratings yet
• Digital PLL

  

  Document6 pages

  Digital PLL

  Peter Podhoranský

  100% (1)
• Analysis and Design of Linear Circuits Solutions

  

  Document49 pages

  Analysis and Design of Linear Circuits Solutions

  Sherif

  No ratings yet
• 2 4 The Smith Chart Package

  

  Document66 pages

  2 4 The Smith Chart Package

  Rodz Gier Jr

  No ratings yet
• Chap10 Phase Locked Loops

  

  Document26 pages

  Chap10 Phase Locked Loops

  Alaa Tobasi

  No ratings yet
• Sedra Cap8

  

  Document79 pages

  Sedra Cap8

  Gustavo Omar Mesones Málaga

  No ratings yet
• Analog Communication Notes

  

  Document73 pages

  Analog Communication Notes

  Aravindha Bhat

  100% (1)
• Ads

  

  Document16 pages

  Ads

  Syed Masudur Rahim

  No ratings yet
• Lecture3 & 4 - Amplitude Modulation - Done

  

  Document73 pages

  Lecture3 & 4 - Amplitude Modulation - Done

  rizwanahmed06

  No ratings yet
• Electronic Circuits Lab Manual 13-12-11 PDF

  

  Document54 pages

  Electronic Circuits Lab Manual 13-12-11 PDF

  REMALLI Keerthi

  No ratings yet
• Power Divider, Combiner and Coupler

  

  Document60 pages

  Power Divider, Combiner and Coupler

  gaurav_juneja_4

  No ratings yet
• Signals & Systems

  

  Document19 pages

  Signals & Systems

  Manish Kushwaha

  No ratings yet
• ECE 308 Exam

  

  Document1 page

  ECE 308 Exam

  chi2king

  No ratings yet
• Satellite Communication Notes

  

  Document50 pages

  Satellite Communication Notes

  Preetham Saigal

  No ratings yet
• 16 - Communication

  

  Document89 pages

  16 - Communication

  Kamran Khursheed

  No ratings yet
• Basic Electrical

  

  Document65 pages

  Basic Electrical

  api-3813463

  100% (1)
• Micro Controller 8051 - New

  

  Document114 pages

  Micro Controller 8051 - New

  gurudyal

  100% (1)
• CC

  

  Document4 pages

  CC

  Suman Kundu

  No ratings yet
• Micro Based Projects

  

  Document12 pages

  Micro Based Projects

  nirveshdagar

  No ratings yet
• Electronics and Communications For Scientists and Engineers

  

  Document417 pages

  Electronics and Communications For Scientists and Engineers

  nirveshdagar

  100% (3)
• Electronic Instrumentation 2nd Edition by P.P.L. Regtien

  

  Document413 pages

  Electronic Instrumentation 2nd Edition by P.P.L. Regtien

  sandeepuae

  100% (7)
• Format For Resume

  

  Document2 pages

  Format For Resume

  api-19625047

  No ratings yet
• Resume Format For All

  

  Document3 pages

  Resume Format For All

  nirveshdagar

  No ratings yet
• Hands On Electronics

  

  Document228 pages

  Hands On Electronics

  api-17233982

  100% (12)
• The 8051 Micro Controller

  

  Document141 pages

  The 8051 Micro Controller

  shilpisabhikhi

  No ratings yet
• Micro Controllers Fundamentals For Engineers and Scientists

  

  Document127 pages

  Micro Controllers Fundamentals For Engineers and Scientists

  jfg8721

  100% (2)
• Basic Electronics V

  

  Document108 pages

  Basic Electronics V

  shilpisabhikhi

  No ratings yet
• Digital Principles Switching

  

  Document545 pages

  Digital Principles Switching

  Harsha

  No ratings yet
• Resume Format

  

  Document2 pages

  Resume Format

  rishikesh1

  100% (3)
• 6th Central Pay Commission Salary Calculator

  

  Document15 pages

  6th Central Pay Commission Salary Calculator

  rakhonde

  100% (436)
• Resume Format 2

  

  Document3 pages

  Resume Format 2

  api-3802379

  100% (4)
• Resume Format

  

  Document2 pages

  Resume Format

  kanthariyahardik

  100% (2)
• Elint 200

  

  Document272 pages

  Elint 200

  nguyenminhgiap

  No ratings yet
• Employment News/Rozgar Samachar

  

  Document2 pages

  Employment News/Rozgar Samachar

  nirveshdagar

  No ratings yet